irclog2html for #asterisk on 20070224

`00:00.04`	`EmleyMoor`	`Cheaper on VoIP for me during daytime`
`00:00.07`	`delmar`	`Bananaskin, this seems the case in many places these days.`
`00:00.35`	`delmar`	`anyway.. does anyone have a solution for me with regards spandsp ? i just need to get my hands on it and the site is down.. maybe someone can email me, or upload to somewhere ?`
`00:01.10`	`delmar`	`EmleyMoor, let me tell you what settings i have on my tdm400 ...`
`00:01.34`	`delmar`	`EmleyMoor, obviously echocancel=yes`
`00:01.56`	`delmar`	`EmleyMoor, echocancelwhenbridged=yes`
`00:02.00`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`00:02.05`	`delmar`	`EmleyMoor, echotraining=800`
`00:02.08`	`EmleyMoor`	`I found that makes matters worse`
`00:02.15`	`Bananaskin`	`hmmm, I was advised to disabled whenbridged`
`00:02.42`	`delmar`	`EmleyMoor, did you modify the source and change the echo canceler or leave the code as default?`
`00:02.45`	`tzanger`	`woo`
`00:03.22`	`delmar`	`i also have..`
`00:03.23`	`delmar`	`rxgain=8.5`
`00:03.23`	`EmleyMoor`	`delmar: I did not alter that part of the code`
`00:03.23`	`delmar`	`txgain=3.5`
`00:03.41`	`InHisName`	`what should I expect from testfeatures item in the features.conf file? I changed the code from #9 to #29 but I seem to have no response.`
`00:04.08`	`delmar`	`EmleyMoor, ok. There are other echocans you can try, but usually the TXgain directly effects the amount of self echo you hear.`
`00:04.25`	`delmar`	`the lower the TXgain the better, but not so low that the other party can't hear you.`
`00:04.59`	`EmleyMoor`	`I've just turned txgain down to -3`
`00:05.01`	`delmar`	`last time I played with spandsp, i found i needed to mess with the rx/tx to get fax send/receive working..`
`00:05.07`	`delmar`	`but then.. the trade off was.. increased echo`
`00:05.10`	`delmar`	`so u cant win :P`
`00:05.34`	`delmar`	`EmleyMoor, have you used the fxotune utility?`
`00:06.02`	`EmleyMoor`	`delmar: Yes, it ran very quickly though so I am not sure`
`00:06.08`	`elriah`	`Anyone here use Vitelity? I wroke a pretty extensive PHP class for their web API if anyone needs it.`
`00:06.11`	`delmar`	`EmleyMoor, how quick?`
`00:06.19`	`EmleyMoor`	`Near-instant`
`00:06.23`	`Bananaskin`	`lol`
`00:06.34`	`delmar`	`EmleyMoor, not good.`
`00:06.45`	`delmar`	`EmleyMoor, it takes a minute or so for each FXO`
`00:06.55`	`EmleyMoor`	`OK, how do I make it work?`
`00:07.55`	`delmar`	`EmleyMoor, http://www.voip-info.org/wiki/view/Asterisk+fxotune`
`00:08.03`	`EmleyMoor`	`I'm doing it now`
`00:08.03`	`*** join/#asterisk ManxPower (n=manxpowe@6.sub-75-201-2.myvzw.com)`
`00:08.09`	`EmleyMoor`	`It's working this time`
`00:08.43`	`EmleyMoor`	`Once I've tuned it, do I have to take action to keep the tuning or is it self-remembering?`
`00:09.21`	`delmar`	`EmleyMoor, if u reload your modules at all, you need to do fxotun -s /etc/fxotune.conf`
`00:09.29`	`*** join/#asterisk psyferre (n=psyferre@host-prestigemag-105-10.customer.ntelos.net)`
`00:09.40`	`delmar`	`fxotune -s /etc/fxotune.conf *`
`00:10.40`	`psyferre`	`hey, folks... anyone have a moment to help with a new installation issue? I'm SO very close to having this done and working, but for some reason I can't get the IVR to answer incoming calls, or outgoing calls to get anything but a busy signal ! I'm sure everything else will fall into place if I can figure that out`
`00:11.07`	`EmleyMoor`	`It seems to have been running for several minutes now`
`00:11.08`	`delmar`	`the echo might be on your FXS more than the FXO.. you need to test them separately. I suggest you use a SIP client .. either a SIP hardphone or a SIP softphone.. when testing the FXO and FXS ports.`
`00:11.17`	`Ryushin`	`Can I use zaptel-1.4 with asterisk-1.2?`
`00:11.18`	`delmar`	`EmleyMoor, yes it does take a long time`
`00:11.29`	`EmleyMoor`	`SIP-FXO echoes worse!`
`00:11.35`	`elriah`	`psyferre: Pastebin your extensions.conf`
`00:11.44`	`EmleyMoor`	`"about a minute per FXO"?`
`00:11.54`	`delmar`	`EmleyMoor, even longer`
`00:12.02`	`EmleyMoor`	`Ah`
`00:12.30`	`EmleyMoor`	`Does it not give a progress indication?`
`00:12.39`	`psyferre`	`http://rafb.net/p/3ByvFN60.html`
`00:12.40`	`Bananaskin`	`no`
`00:12.45`	`*** join/#asterisk diclophis-work (n=jbardin@65.203.37.58)`
`00:12.54`	`diclophis-work`	`are there notions of Arrays in asterisk variable types?`
`00:13.08`	`Carp1`	`if there a feature in asterisk to pick up another extension? Like if the phone is ringing in the next office over and I hear it ringing, I want to pick up my phone and dial like 6XXX where XXX is the extension in the office next door...Say it was 200, I would dial 6200 and answer that call?`
`00:13.09`	`ManxPower`	`diclophis-work: No, but you can emulate it. Want an example?`
`00:13.09`	`*** join/#asterisk lwh (n=lwh192@rdsl-0270.tor.pathcom.com)`
`00:13.19`	`russellb`	`there is an ARRAY function somewhere`
`00:13.26`	`ManxPower`	`Carp1: yes, it is called call pickup`
`00:13.40`	`ManxPower`	`russellb: 1.2 or 1.4?`
`00:13.40`	`russellb`	`might be in 1.4 ... might be just trunk`
`00:13.40`	`russellb`	`not 1.2`
`00:13.50`	`ManxPower`	`I emulate simple arrays in 1.2`
`00:13.56`	`russellb`	`neat`
`00:14.11`	`Carp1`	`Thanks.`
`00:14.54`	`delmar`	`my Asterisk and zaptel here is a bit old and i see new features in the fxotune app, so im gonna go upgrade mine.`
`00:15.29`	`psyferre`	`hmm.... outoing calls give me an "all circuits busy" message instead of a busy signal now`
`00:16.19`	`Carp1`	`With call pickup, can I define the ext I want to pick up? I dont want to just pickup some random extresion`
`00:16.23`	`elriah`	`psyferre: You connected to POTS or IP trunks?`
`00:16.29`	`diclophis-work`	`ManxPower: please`
`00:16.56`	`ManxPower`	`diclophis-work: http://www.fnords.org/~eric/array-example.txt`
`00:17.04`	`psyferre`	`elriah, POTS if i'm not mistaken`
`00:17.05`	`ManxPower`	`pay attention to the INDEX variable`
`00:17.17`	`diclophis-work`	`awesome`
`00:17.24`	`elriah`	`psyferre: Did you pastebin your extensions.conf?`
`00:17.24`	`diclophis-work`	`thats exactlyu what i was gonna try to implement too`
`00:17.32`	`psyferre`	`elriah: yes, http://rafb.net/p/3ByvFN60.html`
`00:17.43`	`elriah`	`jas`
`00:17.48`	`psyferre`	`sorry, didn't put your name in front of it up there, sorry :)`
`00:17.57`	`elriah`	`Is it FreePBX as the pastbin implies?`
`00:18.06`	`psyferre`	`yes.. installed trixbox`
`00:18.21`	`russellb`	`ManxPower: that's cool`
`00:18.23`	`elriah`	`Try #tribox or #freepbx for help on these, I'm not familiar with them (sorry) ...`
`00:18.38`	`psyferre`	`okay, thank you for your time :)`
`00:18.42`	`ManxPower`	`russellb: it's %100 cosmetic, of course. Not REAL arrays`
`00:19.15`	`russellb`	`right`
`00:19.16`	`ManxPower`	`diclophis-work: if you are using 1.4 I would suggest using it's built in real array stuff.`
`00:19.30`	`diclophis-work`	`ManxPower: i am using 1.4`
`00:20.11`	`Grnd-Wire`	`Does anyone know anything about voxee ?`
`00:20.22`	`ManxPower`	`diclophis-work: then you can use my fake arrays or the built in 1.4 real arrays`
`00:20.35`	`*** part/#asterisk psyferre (n=psyferre@host-prestigemag-105-10.customer.ntelos.net)`
`00:20.46`	`diclophis-work`	`ManxPower: do you have an example of real arrays, the wiki only shows how to set them`
`00:21.06`	`*** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com)`
`00:21.08`	`*** join/#asterisk backblue (n=moo@87-196-2-1.net.novis.pt)`
`00:22.59`	`EmleyMoor`	`Which side of the gain do the parameters affect?`
`00:23.44`	`ManxPower`	`diclophis-work: no. I won't use 1.4 until I consider it "stable", and that won't happen for a while.`
`00:24.05`	`EmleyMoor`	`For exa mple, does the RX side get affected by txgain or rxgain?`
`00:24.32`	`ManxPower`	`rx is set using rxgain for that port`
`00:24.55`	`EmleyMoor`	`Hmmmm...`
`00:25.03`	`EmleyMoor`	`Mine seems to be working backwards`
`00:25.09`	`ManxPower`	`Asterisk gets stable enough for my use around the time the next release comes out.`
`00:25.15`	`delmar`	`EmleyMoor, if you increase the TXgain you are increasing the dB level of the audio sent out the FXO to the calling party, hence increasing the ech heard by you on the SIP end.`
`00:25.26`	`ManxPower`	`EmleyMoor: easy enough to get confused.`
`00:26.00`	`EmleyMoor`	`So, if the peak level is not high enough on the rx side, do I do the txgain?`
`00:26.09`	`EmleyMoor`	`(or the rxgain?)`
`00:26.09`	`delmar`	`EmleyMoor, increasing the RXgain therefore, will increase the volume of the caller, making them louder to you on the extension`
`00:26.10`	`ManxPower`	`EmleyMoor: imagine this: rxgain=10 txgain=-5 fxoks=1 fxsks=2 Now what is the gain for a call to the pstn from the fxs thru the fxo?`
`00:27.08`	`EmleyMoor`	`I need this in reality`
`00:27.32`	`delmar`	`does anyone here have a recent version of SpanDSP and the asterisk patch? i dont have it anymore and the main site is down.`
`00:27.42`	`ManxPower`	`well the rxgain for the fxs is 10 and the rxgain for the fxo is 10 so the total gain would be 20`
`00:27.52`	`ManxPower`	`sorry, that is wrong`
`00:28.10`	`*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)`
`00:28.33`	`ManxPower`	`the rxgain for the fxs is 10 and the txgain for the fxo is -5 so your total gain caller -> pstn is 5`
`00:29.14`	`Carp1`	`Calls are coming in to asterisk via NuFone, but then I get a respons on the call saying "The user you are trying to reach is currently unreachable....`
`00:29.17`	`EmleyMoor`	`ManxPower: Consider this: I have an FXO with which I am having problems tuning the audio. fxsen are irrelevant at this stage`
`00:29.20`	`ManxPower`	`EmleyMoor: USUALLY you only need to fix the gains on the FXO, so you would want rxgain=10 txgain=-5 fxoks=1 rxgain=0 txgain=0 fxsks=2`
`00:29.20`	`Carp1`	`http://pastebin.ca/369717`
`00:29.23`	`EmleyMoor`	`No matter what I do, the rx is too week`
`00:29.26`	`EmleyMoor`	`weak`
`00:30.43`	`ManxPower`	`Carp1: last time I used nufone the [nufone] section name must be EXACTLY as nufone says it should be including the correct caps`
`00:30.47`	`*** join/#asterisk backblue (n=moo@87-196-2-1.net.novis.pt)`
`00:31.30`	`Carp1`	`I have NuFone`
`00:31.47`	`EmleyMoor`	`TX is all over the place, RX is just plain too low`
`00:32.15`	`Carp1`	`ManxPower: I believe my caps are in the right places.`
`00:32.36`	`*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)`
`00:32.57`	`ManxPower`	`Carp1: looks like Nufone switched to SER, so my experience is not valid.`
`00:33.05`	`ManxPower`	`Carp1: your paste looks like a registration not a call.`
`00:33.20`	`Carp1`	`Thats what happens when I try to call in.`
`00:33.34`	`Carp1`	`See at the bottom is says destroying call?`
`00:34.16`	`Carp1`	`I would ask their support if it didnt suck...`
`00:34.18`	`diclophis-work`	`i can wildcard #include right?`
`00:34.33`	`wunderkin`	`everything in sip is a call`
`00:34.34`	`ManxPower`	`Carp1: a registration is considered a call`
`00:34.56`	`*** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)`
`00:35.01`	`ManxPower`	`CSeq: 108 REGISTER means it is a register not an INVITE`
`00:35.20`	`Carp1`	`On NuFone member portal, there is a tutorial for outgoing calls...not incoming though :(`
`00:35.23`	`ManxPower`	`diclophis-work: I believe so`
`00:35.28`	`diclophis-work`	`awesome`
`00:37.18`	`EmleyMoor`	`Can setting txgain too high cause silence to come back from the line?`
`00:41.22`	`*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)`
`00:45.14`	`*** join/#asterisk stridernzl (n=neville@125-237-114-232.jetstream.xtra.co.nz)`
`00:45.23`	`*** join/#asterisk Fr0zen_ (i=Fr0zen_@c-67-175-92-171.hsd1.il.comcast.net)`
`00:47.49`	`EmleyMoor`	`Anyone here got distinctive ring detection and caller ID working successfully together in the UK?`
`00:48.20`	`ManxPower`	`Does anyone have a user reference for Asterisk voicemail?`
`00:48.22`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`00:48.22`	`*** mode/#asterisk [+o mog] by ChanServ`
`00:48.44`	`Grnd-Wire`	`ManxPower: There's a decent one on VoIP info..`
`00:48.59`	`Carp1`	`So is anyone successfully recieving inbound NuFone calls? :)`
`00:50.11`	`EmleyMoor`	`I tried the patch many people in Australia found worked, and it doesn't work :-(`
`00:50.37`	`*** join/#asterisk irq (n=dan@wsip-70-167-112-5.sd.sd.cox.net)`
`00:53.26`	`*** join/#asterisk olsen (n=diego@200.61.236.33)`
`00:55.35`	`ManxPower`	`Grnd-Wire: I don't see it. I see many config guides, but no simple, easy 1 page document to give to users that have trouble tieing their shoelaces`
`00:55.37`	`*** join/#asterisk notoriousrab (n=robert_m@207.47.34.74.static.nextweb.net)`
`00:55.42`	`*** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com)`
`00:56.00`	`*** join/#asterisk kuku5 (n=kuku5@c-71-201-219-72.hsd1.il.comcast.net)`
`00:56.14`	`notoriousrab`	`hey, anyone know how to reset a linksys ATA to factory settings, i have been locked out - password does not work`
`00:56.20`	`kuku5`	`Which OS should I use for 1.4 ? I'm trying to decide between centos and redhat`
`00:56.31`	`ManxPower`	`kuku5: which one do you like best?`
`00:57.01`	`ManxPower`	`no matter which distro you pick, many people will tell you not to use it.`
`00:57.05`	`kuku5`	`:)`
`00:57.16`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`00:57.22`	`kuku5`	`I want to use one and re use it :)`
`00:57.39`	`kuku5`	`I'm not a heavy user, so its hard to say which one I like. I don't like rpm's`
`00:57.52`	`kuku5`	`I like to compile things from scratch`
`00:58.09`	`*** part/#asterisk Bobthehunter (n=Bobthehu@145-27.mc.cite.net)`
`00:58.40`	`JT`	`kuku5: debian or gentoo`
`00:59.19`	`kuku5`	`hm`
`00:59.34`	`EmleyMoor`	`I think, apart from putting some of the wiring right and getting my wall phone up, I'm about done`
`01:00.57`	`kuku5`	`is centos basically a free version of redhat enterprise?`
`01:01.18`	`JT`	`yes`
`01:01.21`	`JT`	`repackaged`
`01:02.54`	`kuku5`	`so that is not better than debian ?`
`01:03.02`	`JT`	`imho, no`
`01:03.05`	`JT`	`it's rpm based`
`01:03.12`	`JT`	`and i don't like it much :)`
`01:05.12`	`kuku5`	`centos is rpm based?`
`01:08.41`	`elriah`	`centos is redhat enterprise linux without the redhat commercial software.`
`01:09.02`	`JT`	`rpm = REDHAT Package Manager`
`01:09.06`	`elriah`	`Ubuntu LTS 6.06 Server Edition is my favorite for Asterisk installs.`
`01:09.19`	`*** join/#asterisk topping (n=topping@209-204-141-95.dsl.static.sonic.net)`
`01:11.15`	`Strom_C`	`JT: s/REDHAT/Ridiculous/`
`01:12.17`	`JT`	`:)`
`01:12.35`	`*** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com)`
`01:12.40`	`grinsbalu`	`can someone help me with asterisk and sccp? i've installed asterisk out of the svn from asterisk.org and want to install chan_sccp2 from http://chan-sccp.berlios.de/ but getting these errors.. :/ http://rafb.net/p/AEvBjT73.html`
`01:13.17`	`*** join/#asterisk thoughtpolice (n=austin@ip70-185-140-61.lu.dl.cox.net)`
`01:13.24`	`elriah`	`grinsbalu: Is there a specific reason you're not using SIP?`
`01:13.24`	`grinsbalu`	`asterisk 1.4`
`01:13.29`	`grinsbalu`	`yes`
`01:13.33`	`grinsbalu`	`cisco sccp phone`
`01:13.46`	`elriah`	`Ahh. Which phone? 79x1 or 79x0?`
`01:13.46`	`EmleyMoor`	`grinsbalu: Onw`
`01:13.53`	`grinsbalu`	`1`
`01:13.57`	`grinsbalu`	`no 0`
`01:14.11`	`elriah`	`I have the latest SIP firmware if you want it. It's pretty easy to upgrade.`
`01:14.14`	`grinsbalu`	`no 1`
`01:14.15`	`grinsbalu`	`:D`
`01:14.23`	`grinsbalu`	`oh`
`01:14.29`	`grinsbalu`	`7941`
`01:14.44`	`grinsbalu`	`that would be nice too`
`01:14.46`	`elriah`	`That's what we have, a bunch of 7941's. Doesn't work with NAT without help, though.`
`01:14.52`	`elriah`	`Dunno if skinny does.`
`01:15.08`	`grinsbalu`	`thx alot`
`01:15.22`	`grinsbalu`	`:D`
`01:16.14`	`elriah`	`Sent.`
`01:17.20`	`elriah`	`Just fire up a TFTP server, change the SEP<mac>.cfg to suite you (and rename it with your phone(s) mac address, cycle the power on the phone while holding down #, when the lights blink enter 123456789*0#, and it will update. Also, send option 66 or 150 in your dhcp to point to the IP of your tftpd.`
`01:17.43`	`elriah`	`Altering the XMLDefault.cnf.xml might help you, I didn't need to, just the phone specific file.`
`01:18.43`	`elriah`	`Oh, in Asterisk 1.2.x-1.4.0, you'll need to alter chan_sip.c and take out the text "(0/0)" without the quotes and recompile that module and install it.`
`01:18.48`	`elriah`	`To get MWI to work.`
`01:19.20`	`grinsbalu`	`nice`
`01:19.21`	`grinsbalu`	`thx`
`01:19.50`	`elriah`	`I like the phone a lot, but it doesn't like NAT.`
`01:20.03`	`elriah`	`The quality, though, is what you would typically expect from Cisco.`
`01:20.29`	`elriah`	`Oh, and the ringtones.xml is easy, just drop your PCM or WAVs in the tftpd dir and point to them in ringtones.xml`
`01:21.01`	`elriah`	`As a joke today, we replaced everybody's ring with ducks quacking. Mostly everyone got a kick out of it, mostly.`
`01:22.06`	`diclophis-work`	`how would i play multiple fliles during a Read() ?`
`01:22.14`	`grinsbalu`	`lol`
`01:22.14`	`grinsbalu`	`nice`
`01:22.30`	`grinsbalu`	`back to watching futurama`
`01:22.31`	`grinsbalu`	`:D`
`01:22.43`	`grinsbalu`	`and waiting for the mail to arrive ;)`
`01:26.16`	*** join/#asterisk ez` (n=ez@c66.110.149-45.clta.globetrotter.net)
`01:29.14`	`diclophis-work`	`arg, its not possible to play multiple files during a Read`
`01:29.21`	`diclophis-work`	`how would i make a multi file prompt`
`01:29.36`	`*** join/#asterisk topping (n=topping@adsl-68-122-119-108.dsl.pltn13.pacbell.net)`
`01:30.26`	`*** join/#asterisk adker (n=chatzill@74-33-198-79.br1.glv.ny.frontiernet.net)`
`01:31.44`	`*** join/#asterisk \|dennis\| (n=dennis@200.32.233.82)`
`01:35.14`	`grinsbalu`	`well goin home`
`01:35.16`	`grinsbalu`	`nn`
`01:35.48`	`*** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no)`
`01:37.16`	`*** join/#asterisk genz (n=chatzill@24-197-228-2.static.stpt.wi.charter.com)`
`01:39.56`	`*** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)`
`01:40.03`	`*** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)`
`01:40.42`	`Carp1`	`Does anyone know how to add 3 digit dialing to my digit map on a Polycom 501?`
`01:45.39`	`ManxPower`	`Carp1: http://www.fnords.org/~eric/polycom-config-examples`
`01:46.05`	`*** join/#asterisk genz_ (n=chatzill@im.jobdig.com)`
`01:49.08`	`*** join/#asterisk apardo (n=apardo@87.217.145.9)`
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`02:10.13`	`Zilasb`	`hello`
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`02:11.04`	`Zilasb`	`I have serverA and serverB. Sip clients on both servers. How can A know about register information of sip user on B?`
`02:12.02`	`Zilasb`	`Is it possible?`
`02:14.18`	`ManxPower`	`Zilasb: no.`
`02:14.35`	`ManxPower`	`a calls b via b's server. b's server then does what it does`
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`02:17.10`	`Zilasb`	`So there no way A sip registered user to call B sip user locally?`
`02:18.11`	`Zilasb`	`ManxPower: what I was thinking I could use same extension.conf file for both servers`
`02:18.32`	`Carp1`	`ManxPower: I dont understand those examples? How to I edit a.cfg file? I have web based editing???`
`02:19.20`	`Zilasb`	`ManxPower: and I was thinking if there is any other way from B server to bypass all the registry information to A... Would SER help me?`
`02:22.33`	`ManxPower`	`Carp1: if you cannot edit a config file then you need to step away from the computer and find another hobby`
`02:23.10`	`ManxPower`	`Carp1: you can look at the dial pattern section of the sip.cfg, paste that in your polycom "my first phone" web interface.`
`02:23.29`	`ManxPower`	`Zilasb: I know nothing about SER`
`02:25.36`	`Zilasb`	`my thing is that I have 2 servers in other parts of the wolrd using literally same config files. I just need some kind of solution to be able to make calls in between sip users registered to other servers. Nothing comes to my head how to implement that`
`02:26.21`	`backblue`	`Zilasb: dundi?`
`02:28.50`	`ManxPower`	`Carp1: if you want to do anything complicated with the polycoms you will have to set up a tftp or ftp servers and have the phone download their configs from the server.`
`02:30.23`	`clyrrad`	`ManxPower: they support https too?`
`02:32.29`	`backblue`	`yes`
`02:32.39`	`backblue`	`http,https,ftp,tftp`
`02:32.47`	`backblue`	`and more i think, but i dont remeber`
`02:33.09`	`clyrrad`	`what you think of polycom vs sipura?`
`02:33.29`	`backblue`	`polycom for me, it's the best phone on the market.`
`02:33.40`	`clyrrad`	`why do you like it more than Sipura?`
`02:33.42`	`ManxPower`	`clyrrad: the x01 series with new bootroms do support https`
`02:33.45`	`Carp1`	`ManxPower...I've never had a "real" IP phone before.....Grandstream a few years back....I dont know where the config files are...`
`02:34.12`	`Carp1`	`I goto the web interface, I've been through everything, I only see where you can edit valued via input boxes.`
`02:34.12`	`backblue`	`clyrrad: i was giving my opinion, i dont use sipura.`
`02:34.28`	`clyrrad`	`Carp1: I missed the first part of your conversation, but if im on track with your question, you send the configs into the phones via FTP, TFT, HTTP, or HTTPS`
`02:34.32`	`ManxPower`	`Carp1: they come with the firmware that you should get from polycom (2nd from the most recent versions) or your reseller (newest version)`
`02:34.59`	`backblue`	`clyrrad: no, the phones get them for you.`
`02:35.04`	`Carp1`	`I didnt get anything but a new phone and power cable and cat5 with PoE injector`
`02:35.10`	`Carp1`	`no CD or manuel.`
`02:35.18`	`clyrrad`	`backblue: even with out connection to a provisioning server?`
`02:35.24`	`backblue`	`Carp1: yeah, that's polycom :(`
`02:35.25`	`Carp1`	`Oh, the phone has an FTP server?`
`02:35.48`	`backblue`	`clyrrad: yes, you have to manually configure on the phone`
`02:35.57`	`backblue`	`to get the config from where it should be`
`02:36.06`	`backblue`	`Carp1: no.`
`02:36.15`	`clyrrad`	`backblue: ok but if he connected to a provisioning server it would do it all for him, would it not?`
`02:36.26`	`backblue`	`clyrrad: yes it will.`
`02:36.30`	`*** join/#asterisk r0d3nt (n=RatMan@punk.valuetel.net)`
`02:36.32`	`clyrrad`	`ok cool`
`02:36.38`	`clyrrad`	`so its like the Sipura products then`
`02:36.38`	`backblue`	`And does a very good job.`
`02:36.54`	`backblue`	`yes, all the professional line, of IP phones, do that.`
`02:36.56`	`clyrrad`	`yea I use provisioning server for all our Sipura stuff and it works great`
`02:37.40`	`Carp1`	`Ok....I know I'm sounding like an idiot...but I just dont know lol...Where are the config files located? How do I open them....I only see the easy editing via web interface.`
`02:38.20`	`clyrrad`	`Carp1: generally you can set everyting from the Webadmin`
`02:38.42`	`backblue`	`not in polycom`
`02:38.44`	`Carp1`	`Ok...What I really am looking to do it add 3 digits to the dial pattern`
`02:38.56`	`backblue`	`you will really need remote provisioning`
`02:38.59`	`Carp1`	`11 is already added but I dont understand how to add 3.`
`02:39.15`	`clyrrad`	`backblue: how you config your polycom in this situation?`
`02:39.32`	`backblue`	`for the 3 digits?`
`02:39.43`	`Carp1`	`3 digit extensions`
`02:39.48`	`Carp1`	`so I dont have to hit "send" each time.`
`02:39.57`	`clyrrad`	`for most advanced stuff if you dont have a capable webadmin`
`02:40.22`	`clyrrad`	`is it that you use some kind of profile compiler and sync it with the device?`
`02:40.29`	`elriah`	`Carp1: Are you selecting a line first or dialing the number first?`
`02:40.36`	`Zilasb`	`Hey one tech question. How many simultenious channels asterisk can handle?`
`02:40.45`	`ManxPower`	`Zilasb: 1024`
`02:40.51`	`clyrrad`	`Zilasb: 2048`
`02:40.52`	`Corydon76-home`	`As many as the hardware will allow`
`02:40.53`	`clyrrad`	`:p`
`02:40.53`	`Carp1`	`I am opening up a line`
`02:40.57`	`Zilasb`	`?`
`02:40.58`	`backblue`	`clyrrad: i just manually edit the config, and reload the phone.`
`02:40.59`	`ManxPower`	`unleess you want a REAL answer`
`02:41.02`	`Carp1`	`get dialtone, and then dial 3 digits`
`02:41.19`	`Corydon76-home`	`Your limit is based upon CPU, available memory, and available bandwidth`
`02:41.19`	`ManxPower`	`Zilasb: the answer is "it depends"`
`02:41.21`	`elriah`	`Are you provisioning via ftp/tftp or are you manually configuring phones?`
`02:41.31`	`ManxPower`	`elriah: he doesn't know how to do that.`
`02:41.37`	`clyrrad`	`backblue: yea but where are you editing the config if not in webadmin?`
`02:41.54`	`backblue`	`Carp1: and if you wait after sending the 3 digits, it does never dial?`
`02:42.02`	`Carp1`	`no.`
`02:42.05`	`elriah`	`Carp1: You're configuring the phones how? With the menus on the phone?`
`02:42.13`	`backblue`	`clyrrad: locally on the provisioning server.`
`02:42.13`	`ManxPower`	`clyrrad: you edit the text config file on the file server. Or you config the phone via the web interface on the phone.`
`02:42.15`	`Zilasb`	`ManxPower: depends on hardware right?`
`02:42.37`	`ManxPower`	`Zilasb: hardware, codecs, technology, interfaces, protocols`
`02:42.51`	`elriah`	`Carp1: HOW are you configuring the phones, pick one: phone interface, web interface, ftp/tftp/http?`
`02:42.55`	`ManxPower`	`Carp1: you can't just "enable 3 digit dialing" and expect it to work.`
`02:42.56`	`Carp1`	`No, with the web interface`
`02:42.58`	`clyrrad`	`ManxPower: yea thats what I thought but backblue said the webadmin was limited - unless I mis-understood him`
`02:43.10`	`elriah`	`Paste your current dial plan from the web interface.`
`02:43.15`	`Carp1`	`no, i dial 3 digits, and it never goes throuigh until i hit send`
`02:43.24`	`elriah`	`Carp1: Paste your current dial plan from the web interface.`
`02:43.25`	`ManxPower`	`Carp1: you need to design your dialplan so that you have no overlapping extension patterns`
`02:43.50`	`ManxPower`	`clyrrad: you can set the dialplan via the web interface. yes, the web interface is very limited.`
`02:43.53`	`elriah`	`Or, just add \|xxx\| to the end of it.`
`02:43.55`	`backblue`	`yes, webadmin it's limited, you cant do everything in the webadmin.`
`02:44.09`	`elriah`	`You can specify a dial pattern in the web admin.`
`02:44.29`	`elriah`	`Carp1: Just add \|xxx\| to the end of your dial pattern, being sure not to duplicate vertical bars.`
`02:44.32`	`Carp1`	`ahh, I've had enough for tonight...Im goin to the bar..,thanks everyone`
`02:44.34`	`clyrrad`	`ManxPower: k so my last part of that question was how you best config them.... for advanced reasons... you make a config.txt or something similar... then sync it with the phone?`
`02:44.45`	`Carp1`	`I might have to pay someone on monday to help me learn about this Polycom.`
`02:44.46`	`elriah`	`Carp1: I gave you the answer dude. DID YOU SEE IT?`
`02:44.51`	`elriah`	`Dear God.`
`02:44.55`	`ManxPower`	`clyrrad: the configs are XML and the default config files come with the firmware`
`02:44.57`	`Carp1`	`Oh`
`02:44.59`	`Carp1`	`I didnt`
`02:45.00`	`Carp1`	`but thanks`
`02:45.09`	`clyrrad`	`ManxPower: ok so thats like Sipura hardware then`
`02:45.14`	`elriah`	`Carp1: Just read the book, it's pretty clear.`
`02:45.22`	`elriah`	`Carp1: Or the info at voip-info.org`
`02:45.43`	`Carp1`	`I dont have the book lol.`
`02:45.53`	`Carp1`	`I know, I can probably find it on the polycom website.`
`02:46.00`	`clyrrad`	`ManxPower: What you prefer Polycom or Sipura?`
`02:46.01`	`elriah`	`Carp1: You can download it from polycom.com like 99% of manuals these days.`
`02:46.03`	`backblue`	`probably you dont! :)`
`02:46.07`	`ManxPower`	`elriah: he got the phone from some fly by night reseller`
`02:46.21`	`backblue`	`elriah: only if it uses old versions`
`02:46.32`	`ManxPower`	`clyrrad: I manage 100+ polycom phones and no other brand of phone. That should be your answer.`
`02:46.35`	`backblue`	`polycom only gives stuff to resellers or something`
`02:46.46`	`clyrrad`	`ManxPower: indeed`
`02:47.03`	`elriah`	`backblue: You can get the firmware if you search for it for a few minutes.`
`02:47.11`	`*** join/#asterisk linagee (n=linagee@unaffiliated/linagee)`
`02:47.12`	`elriah`	`backblue: Still doesn't change the fact that the manual explains all.`
`02:47.16`	`backblue`	`elriah: so, download me version 2.*`
`02:47.19`	`*** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner)`
`02:47.19`	`elriah`	`backblue: And you can download back versions.`
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`02:48.10`	`ManxPower`	`backblue: you can only download old releases, but they are fine for our use. I am a certified polycom tech and so get access to all the firmwares`
`02:48.27`	`ManxPower`	`and we don't use 2.x, we still use 1.6.7`
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`02:48.29`	`*** mode/#asterisk [+o denon] by ChanServ`
`02:48.59`	`elriah`	`Polycom will give you the new firmware if you call them and ask nicely. And it's available online if you search in any number of places.`
`02:49.01`	`backblue`	`ManxPower: i will be next month or so too, but until there, have to ask them to a reseller!`
`02:49.05`	`backblue`	`ManxPower: why dont use 1.6.7?`
`02:49.20`	`backblue`	`elriah: that should not be needed!`
`02:49.39`	`backblue`	`elriah: and i will not put firmware from some place on the web`
`02:49.52`	`Zilasb`	`what is interesting when my asterisk box has a 60+ simulteneous calls g729 it takes forever to do a answer on channel... Maybe slow hardware dual p3. Just upgraded to dual p4 will see if any change`
`02:49.54`	`elriah`	`backblue: Well that's a pointless debate... They can do what they want with their software...`
`02:49.58`	`backblue`	`probably you never hear of the backdoors on ciscos IOS and stuff like that.`
`02:50.13`	`backblue`	`elriah: yes.`
`02:50.15`	`elriah`	`backblue: Well, pack the phone back up and put it on ebay.`
`02:50.33`	`elriah`	`backblue: It's useless I guesss.`
`02:50.48`	`backblue`	`elriah: what it is useless?`
`02:51.16`	`ManxPower`	`backblue: it's better than Cisco, which makes you PAY $120 for a legal copy of the SIP firmware`
`02:51.24`	`elriah`	`backblue: Because there is a disconnect between the hardware and the user or something. If you troubleshoot a problem from the point of defeat then you'll never solve it.`
`02:51.24`	`*** join/#asterisk Barmal (i=Zilas_@c-24-99-8-2.hsd1.ga.comcast.net)`
`02:51.58`	`elriah`	`NO THEY DON'T. Cisco firmware is at most $11 which is the cost of a 1-year support contract which entitles you to free updates and all SIP/Skinny firmware.`
`02:52.12`	`ManxPower`	`And no, a Cisco support contract does not give you legal license for the SIP firmware.`
`02:52.18`	`backblue`	`ManxPower: IOS cost much more.`
`02:52.31`	`ManxPower`	`It gives you the legal right for minor point updates of whatever firmware you already have.`
`02:52.36`	`elriah`	`ManxPower: Eh?`
`02:52.40`	`JerJer`	`elriah: that $11 only applies if you acquired the phone via a Cisco authorized reseller`
`02:53.00`	`elriah`	`ManxPower: Even without a contract I called them and they gave me the SIP firmware knowing I had a skinny phone, so I don't guess they really care that much.`
`02:53.00`	`JerJer`	`otherwise you will need to have the phone re-certified by cisco, which costs just about as much as a new fone`
`02:53.02`	`ManxPower`	`elriah: read the fine print of the support contract.`
`02:53.05`	`backblue`	`"disconnect between the hardware and the user" -> dont understand this statement!`
`02:53.16`	`backblue`	`sorry but my english it's not the best.`
`02:53.20`	`elriah`	`I'm out, peace fellas.`
`02:53.25`	`Barmal`	`ManxPower: so there is no way in * to foward all the real time sip registry information from one server to other?`
`02:53.56`	`ManxPower`	`Barmal: not that I am aware of. The ARA ("Realtime") might do it`
`02:54.04`	`backblue`	`Barmal: SER, forget asterisk.`
`02:54.43`	`backblue`	`Barmal: why are you asking a car to bark?`
`02:54.54`	`backblue`	`s/car/cat/g`
`02:55.05`	`backblue`	`hehe, nice.`
`02:55.21`	`Barmal`	`Funny :)`
`02:56.01`	`ManxPower`	`JerJer: I see you switched to SER at NuFone`
`02:56.15`	`JerJer`	`we have always used SER`
`02:56.36`	`JerJer`	`ser+mediaproxy+asterisk+my magic sauce`
`02:56.46`	`Barmal`	`backblue: I was thinking about SER, but all my billing is on AGI... So ser would be as a registar and proxy nothing more???`
`02:56.48`	`ManxPower`	`JerJer: Ah. Maybe I only used IAX when I had NuFone`
`02:56.51`	`backblue`	`JerJer: ser+freeswitch does not fit?`
`02:56.59`	`JerJer`	`backblue: hell no`
`02:57.03`	`ManxPower`	`I miss having an internet connection with less than 900ms latency`
`02:57.15`	`backblue`	`Barmal: ser would do only what you were asking.`
`02:57.17`	`JerJer`	`Barmal: your billing will not scale`
`02:57.25`	`backblue`	`read again what you asking`
`02:57.29`	`backblue`	`JerJer: why not?`
`02:57.42`	`JerJer`	`fork new thread`
`02:57.45`	`JerJer`	`parse script`
`02:57.49`	`JerJer`	`execute script`
`02:57.55`	`JerJer`	`for every call`
`02:58.01`	`backblue`	`that's your billing?`
`02:58.25`	`JerJer`	`backblue: not mine - for damn sure`
`02:58.36`	`backblue`	`well, freeswitch scale with ser`
`02:59.00`	`backblue`	`you have to fit the billing, but a better plataform for shore.`
`02:59.06`	`JerJer`	`um no`
`02:59.17`	`florz`	`JerJer: Where did he say it was a script?`
`02:59.26`	`JerJer`	`florz: AGI`
`02:59.40`	`florz`	`JerJer: Why does an AGI need to be a script?`
`02:59.48`	`Corydon76-home`	`Yeah, AGI is about the worst when it comes to scaling`
`02:59.49`	`JerJer`	`if not script it then has to execute a new process each and every time`
`03:00.03`	`florz`	`JerJer: Which isn't really _that_ expensive ...`
`03:00.04`	`Barmal`	`so if SER is before * and it forwards all sip req to * we are adding additional lantecy?`
`03:00.06`	`Corydon76-home`	`FastAGI is much better, though still less than optimal`
`03:00.08`	`JerJer`	`FastAGI is better, bu t not good`
`03:00.35`	`backblue`	`Barmal: SER it's very good in it's job, it does not do any media stuff, and it's pretty fast.`
`03:00.35`	`JerJer`	`Barmal: SER only deals with SIP signalling`
`03:00.43`	`backblue`	`probably you will not fell anything`
`03:01.28`	`Corydon76-home`	`Barmal: SER assists in setting up the call, but it does not add additional latency`
`03:01.37`	`Barmal`	`I need to read about it more. Whitch one SER or openSER?`
`03:01.41`	`Corydon76-home`	`That's the whole point of a proxy`
`03:01.48`	`Corydon76-home`	`Barmal: either one`
`03:02.10`	`Corydon76-home`	`A gateway, such as Asterisk, is the only endpoint which could possibly add latency`
`03:02.27`	`Corydon76-home`	`but a gateway lets your bridge between media, which a proxy will not let you do`
`03:02.55`	`Barmal`	`I see debian has openser... SER deal with datagrams level not with sip users right?`
`03:03.15`	`Corydon76-home`	`Barmal: yes, it deals with SIP users`
`03:03.21`	`JerJer`	`Barmal: SER is older, less features`
`03:03.27`	`Corydon76-home`	`Barmal: you're oversimplifying`
`03:03.34`	`*** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar)`
`03:03.48`	`JerJer`	`OpenSER is actively being developed, which means bugs and stability problems`
`03:03.56`	`JerJer`	`but it has more features`
`03:04.13`	`Corydon76-home`	`Barmal: there are two different streams involved. The first is the control protocol, which is what the proxy works with. The second is the media stream, which is passed directly between peers, not through the proxy`
`03:05.11`	`Barmal`	`Corydon: Does ser have in configuration something like sip.conf, I mean info about sip user?`
`03:05.27`	`Corydon76-home`	`Barmal: yes`
`03:05.30`	`JerJer`	`Barmal: SER is very radically different than Asterisk`
`03:05.35`	`Corydon76-home`	`Barmal: or, it can anyway`
`03:05.54`	`Corydon76-home`	`Barmal: it can also be blind about peer configurations`
`03:06.41`	`JerJer`	`SER can be setup to very simply route sip messages around`
`03:06.59`	`JerJer`	`or it can statefully inspect the packets and keep track of dialogs and what not`
`03:07.06`	`JerJer`	`all depends on the configuration you give it`
`03:07.27`	`*** join/#asterisk orkid (n=orkid@dataq2.utias.utoronto.ca)`
`03:07.29`	`Barmal`	`the bad part so many books about * but not that many on ser... Anyway sounds like a powerfull tool... I am at the point that I need something more since company and users are growing...`
`03:07.56`	`JerJer`	`Barmal: http://www.jeremy-mcnamara.com/index.php/2007/02/22/seropenser-configuration-wizard/`
`03:08.06`	`Barmal`	`thx`
`03:10.12`	`bkw__`	`any mISDN folks around?`
`03:10.20`	`Barmal`	`1 day old article? Wov`
`03:10.55`	`ManxPower`	`I pretty much avoid all these issues by working in a corporate environment rather than an ITSP environment`
`03:11.10`	`ManxPower`	`<nelson>Ha! Ha!</nelson>`
`03:11.26`	`JerJer`	`ser still is relevant for the enterprise`
`03:11.39`	`JerJer`	`i've done some pretty sweet presence based stuff using SER`
`03:11.40`	`ManxPower`	`JerJer: In what way?`
`03:12.25`	`ManxPower`	`JerJer: My users don't use text messaging on their phones because "its too complicated", but if your users are tech literate, then I can see your point.`
`03:12.44`	`JerJer`	`doens't necessarily have to be text messaging`
`03:13.17`	`JerJer`	`one can detect where a user is logged in at and send them the call there`
`03:13.27`	`JerJer`	`versus blindly sending to all possible devices`
`03:13.31`	`ManxPower`	`My point is that they have enough trouble remembering to dial 9 for an outside line, anything more complicated than that and they won't use it and will complain about it.`
`03:13.44`	`JerJer`	`prolly :)`
`03:13.45`	`*** join/#asterisk PBX-Nick (i=Sktymelb@202.10.94.18)`
`03:13.45`	`ManxPower`	`We have found that the fewer options we give users the less they complain`
`03:14.33`	`*** join/#asterisk joaovianna (i=joaovian@ool-4354d1a8.dyn.optonline.net)`
`03:14.35`	`ManxPower`	`One out of 15 receptionists at the 15 offices even knew what a blind transfer is.`
`03:16.02`	`ManxPower`	`People constantly overestimate the end users willingness to learn anything new.`
`03:16.51`	`ManxPower`	`at least when it comes to telecom`
`03:17.12`	`ManxPower`	`for computers they will spend hours downloading the latest spyware laden screen saver`
`03:17.18`	`JerJer`	`i guess it depends on the enterprise`
`03:17.25`	`ManxPower`	`JerJer: nod`
`03:17.30`	ez`	`asterlink is dead ???`
`03:17.36`	`JerJer`	`seems like most of my customers are totally into features and functions`
`03:17.44`	`ManxPower`	`JerJer: I envy you then`
`03:17.55`	`ManxPower`	`JerJer: but you are a service provider, they want your service.`
`03:18.02`	`JerJer`	`yup`
`03:18.18`	`ManxPower`	`my users put up with IT because they can't figure out how to fire the entire department and put their nephew in charge of it.`
`03:18.18`	`Barmal`	`is it a big differnce in configuration files between ser and openser?`
`03:19.09`	`ManxPower`	`(and yes, we had several attempts to do EXACTLY that)`
`03:19.12`	`JerJer`	`Barmal: very ver much so`
`03:19.14`	`JerJer`	`+y`
`03:20.32`	`bkw__`	ez`, what?
`03:20.34`	`*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)`
`03:21.31`	`joaovianna`	`Anyone using asterisk with video ?`
`03:21.38`	ez`	`hum ... ; dunno; website seem unreachable ...`
`03:21.48`	`bkw__`	`dns is a bit slow today`
`03:21.55`	`bkw__`	ez`, what country are you in?
`03:22.03`	ez`	`canada`
`03:22.14`	`bkw__`	ez`, we'll talk in private
`03:24.31`	`JerJer`	`it doesn't work in michigan or Los Angeles either`
`03:25.27`	`*** join/#asterisk \|dennis\| (n=dennis@200.32.233.82)`
`03:26.11`	`Qwell`	`Strom_C: nub`
`03:26.20`	`Strom_C`	`?`
`03:26.22`	`Strom_C`	`is this re my bug?`
`03:26.31`	`Qwell`	`yeah, you caught a 10-20 minute window where it didn't compile`
`03:26.36`	`Qwell`	`svn up :)`
`03:26.41`	`Qwell`	`(on both)`
`03:26.42`	`Strom_C`	`i waited and then svn'ed again a few times :)`
`03:26.47`	`Strom_C`	`so i figured it was afe`
`03:27.00`	`ManxPower`	`mt first choice for a gmail userid was available`
`03:27.10`	`Qwell`	`should've been fixed in 56548`
`03:27.18`	`bkw__`	`now to kill someone`
`03:27.36`	`Qwell`	`Strom_C: update zap too`
`03:27.39`	`Strom_C`	`qwell: hmm ok, a new version of that module didnt download; let me try it`
`03:27.57`	`wunderkin`	`ManxPower, your nickname always makes me think of Max Power`
`03:28.13`	`ManxPower`	`wunderkin: it's supposed to. reference to the Simpsons, of course`
`03:28.17`	`JerJer`	`i think that is intentional`
`03:28.20`	`wunderkin`	`heh`
`03:28.33`	`Qwell`	`ManxPower: Simpsons? What's that?`
`03:28.33`	`wunderkin`	`his name can be said by anyone! la la la`
`03:30.14`	`florz`	`wunderkin: Your nickname always makes me think of some outstanding chin or something ... =:-)`
`03:30.33`	`Strom_C`	`qwell: i'm compiling`
`03:30.51`	`JerJer`	`she has more chin's than a chienese phone book`
`03:32.17`	`wunderkin`	`florz, your nickname always makes me think of... floors..`
`03:32.28`	`Strom_C`	`what about Flooz?`
`03:32.33`	`Strom_C`	`the intarweb currency !!!!!!!!!`
`03:32.41`	`wunderkin`	`floosies?`
`03:32.45`	`JerJer`	`or floosie`
`03:32.45`	`Qwell`	`ugh`
`03:33.23`	`wunderkin`	`ManxPower, but you didn't get your nickname off of a hairdryer..`
`03:33.49`	`florz`	`wunderkin: hehe :-) - but seriously, is there a d or an n missing at the end of your nick? Or is it of some completely different origin?`
`03:34.07`	`wunderkin`	`yeah.. i was stuck with the 9 character limit from undernet and left it there`
`03:34.28`	`*** join/#asterisk coppice (n=chatzill@13.168.17.210.dyn.pacific.net.hk)`
`03:36.28`	`ManxPower`	`wunderkin: I got it from a guy that got it off a hair dryer`
`03:37.37`	`wunderkin`	`the name again is mr plow`
`03:40.33`	`Strom_C`	`qwell: it still crashes`
`03:40.40`	`Qwell`	`suck`
`03:40.54`	`Strom_C`	`i hope you didnt close the bug already`
`03:40.59`	`Qwell`	`nope`
`03:42.49`	`Strom_C`	`if you're interested in poking at it, I can let you into my box`
`03:43.02`	`Qwell`	`should be trivial to reproduce`
`03:43.14`	`Strom_C`	`nod`
`03:43.27`	`Strom_C`	`well, just on the off chance it's something stupid I'm doing`
`03:43.28`	`Qwell`	`EC2?`
`03:43.38`	`Qwell`	`Strom_C: doubtful. Kevin was doing stuff today`
`03:43.43`	`Strom_C`	`ah`
`03:43.59`	`bkw__`	`Qwell Amazon Elastic Compute Cloud`
`03:44.03`	`Qwell`	`oh`
`03:44.27`	`bkw__`	`I love this S3 stuff too`
`03:44.31`	`bkw__`	`nice stuff`
`03:44.38`	`Qwell`	`S3?..`
`03:44.56`	`bkw__`	`Amazon Simple Storage Service`
`03:45.06`	`Qwell`	`ASSS?`
`03:45.12`	`bkw__`	`haha never thought of that`
`03:45.14`	`bkw__`	`but that is funny`
`03:45.24`	`bkw__`	`guess thats why they call it S3`
`03:45.25`	`bkw__`	`haha`
`03:45.29`	`Qwell`	`Amazon Simple Storage Elastic Service`
`03:46.18`	`*** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner)`
`03:51.10`	`*** join/#asterisk bmg505 (n=leon@c1-86-16.rndf.isadsl.co.za)`
`03:54.08`	`*** join/#asterisk AJaymn (n=boiwonde@24-159-236-181.dhcp.mdsn.wi.charter.com)`
`03:54.44`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`03:54.44`	`*** mode/#asterisk [+o mog] by ChanServ`
`03:56.00`	`*** join/#asterisk rudholm (i=rudholmm@nat/yahoo/x-88c0b6ebf58446c4)`
`03:56.28`	`Strom_C`	`i already told you: the answer is balls`
`03:56.42`	`*** join/#asterisk Zaw (i=zaw@unaffiliated/zaw)`
`03:57.04`	`rudholm`	`balls?`
`03:57.08`	`Strom_C`	`yes`
`03:57.29`	`*** join/#asterisk ars247 (n=no@ftw-66-42-87-41.customer.stknca.fire2wire.com)`
`03:57.59`	`AJaymn`	`2 in a sack!`
`03:58.01`	`AJaymn`	`;)`
`03:58.08`	`rudholm`	`as it should be`
`04:00.02`	`*** join/#asterisk ToyMan (n=Stuart@user-12lcqkq.cable.mindspring.com)`
`04:00.05`	`rudholm`	`you and your General Telephone`
`04:00.41`	`rudholm`	`was GLadstone "GTE" when you lived there? or had it already become "Verizon"?`
`04:00.48`	`Strom_C`	`oh, it was GTE`
`04:00.51`	`Strom_C`	`I grew up there`
`04:01.14`	`rudholm`	`but it wasn't "General Telephone", right?`
`04:01.18`	`rudholm`	`you're too young for that`
`04:01.22`	`Strom_C`	`right`
`04:01.37`	`Strom_C`	`I lived on GLadstone from 1983-1994`
`04:01.49`	`Qwell`	`GL(?)adstone?`
`04:01.52`	`rudholm`	`yeah, I remember General Telephone as well`
`04:01.59`	`Strom_C`	`qwell: yep`
`04:02.09`	`Qwell`	`clue in the nub`
`04:02.15`	`rudholm`	`I grew up in ANgelus`
`04:02.20`	`Qwell`	`why the cap L?`
`04:02.24`	`Strom_C`	`http://www.stromcarlson.com/payphones/pcplca001.jpg`
`04:02.28`	`Strom_C`	`that's GLadstone`
`04:02.29`	`joaovianna`	`Anyone using video in asterisk ? I'm using Grandstream 3000 but no sucess using video.`
`04:02.45`	`rudholm`	`well, back in the day, telephone numbers used to be denoted by exchange names`
`04:03.06`	`Qwell`	`oh, right`
`04:03.06`	`*** part/#asterisk Johnnie (n=jdlewis@jdlewis.org)`
`04:03.17`	`rudholm`	`oh, speaking of history, Strom, remember how I was showing you that AAA book from 1919?`
`04:03.19`	`Qwell`	`so, 45X?`
`04:03.26`	`rudholm`	`and it had phone numbers like "Home" this and "Central" that?`
`04:03.56`	`rudholm`	`I recently read something about how the original telephone companies in Los Angeles were The Home Telephone Company and The Central(?) Telephone Company`
`04:03.59`	`rudholm`	`and they didn't interconnect`
`04:04.04`	`rudholm`	`Qwell: yes`
`04:04.08`	`Strom_C`	`brb phone`
`04:04.19`	`rudholm`	`I lived in GLadstone5 for 8 years`
`04:04.29`	`rudholm`	`grew up in ANgelus`
`04:04.39`	`Strom_C`	`and now you're back in ANgelus`
`04:04.44`	`rudholm`	`yes, yes I am`
`04:04.54`	`rudholm`	`WEbster was fun`
`04:04.56`	`Strom_C`	`ok, im back`
`04:05.07`	`Strom_C`	`rudholm: oh interesting re Home and Central`
`04:05.13`	`Strom_C`	`I remember GTE bought...was it Home?`
`04:05.16`	`rudholm`	`yep`
`04:05.28`	`rudholm`	`well, The General Telephone Company bought out a number of companies`
`04:05.30`	`rudholm`	`including Home`
`04:05.47`	`rudholm`	`they had Santa Monica, Long Beach, and some other areas`
`04:05.57`	`Strom_C`	`qwell: I'll give you three guesses what this restaurant's telephone number is`
`04:05.58`	`Strom_C`	`http://www.stromcarlson.com/payphones/pcplca008.jpg`
`04:06.32`	`Qwell`	`56849474, duh :p`
`04:06.38`	`rudholm`	`hehe`
`04:06.38`	`Qwell`	`erm`
`04:06.44`	`Qwell`	`s/9/3/`
`04:06.45`	`Strom_C`	`thats not even a NANP number, silly`
`04:06.53`	`Qwell`	`wait, wtf`
`04:07.04`	`rudholm`	`Qwell: what country are you in?`
`04:07.13`	`rudholm`	`is that a Mexico City number?`
`04:07.18`	`Qwell`	`4543474 :p`
`04:07.18`	`Strom_C`	`rudholm: the people's republic of alabama`
`04:07.21`	`rudholm`	`Paris? Tokyo?`
`04:07.26`	`Qwell`	`had an extra number in there somewhere`
`04:07.57`	`rudholm`	`Good old Sunset Blvd and PCH`
`04:08.08`	`Strom_C`	`what's amusing is that, last time I checked, their website still lists their number as GL4-FISH`
`04:08.12`	`rudholm`	`my car died right there once`
`04:08.17`	`Strom_C`	`oh?`
`04:08.22`	`ManxPower`	`what are you doing with a pic of a place in alabama`
`04:08.41`	`Qwell`	`ManxPower: That is not alabama :P`
`04:08.41`	`rudholm`	`yeah, the battery connector came off and on modern cars, the ignition system dies if that happens (on my older cars, that wasn't an issue)`
`04:08.47`	`*** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au)`
`04:09.01`	`Strom_C`	`gah, they changed the site and now it just says 454`
`04:09.06`	`rudholm`	`booo`
`04:09.08`	`Qwell`	`ManxPower: palm trees in the distance.. many, many powerlines`
`04:09.13`	`ManxPower`	`I need to get to Huntsville next time oej is there`
`04:09.22`	`Qwell`	`I count like 8 stoplights`
`04:09.27`	`Qwell`	`That's so LA :p`
`04:09.29`	`rudholm`	`wait, they don't have power lines in Alabama?`
`04:09.34`	`rudholm`	`I thought they had electricity now`
`04:09.39`	`Qwell`	`rudholm: not that many, heh`
`04:09.42`	`Strom_C`	`rudholm: only downtown`
`04:09.47`	`Qwell`	`oh, and traffic`
`04:09.50`	`rudholm`	`well, that's the beach`
`04:10.07`	`Strom_C`	`qwell: oh please. University in the afternoon is comparable to the 405`
`04:10.15`	`rudholm`	`note the chevy in the right lane going to try to jump ahead when the light turns green`
`04:10.16`	`Qwell`	`nah`
`04:10.24`	`Qwell`	`rudholm: heh`
`04:10.33`	`rudholm`	`three lanes become two right there`
`04:10.40`	`ManxPower`	`so, so who is in AL?`
`04:10.48`	`rudholm`	`Qwell is in AL`
`04:10.54`	`rudholm`	`apparently they have no electricity there`
`04:10.56`	`rudholm`	`or signal lights`
`04:10.57`	`Qwell`	`ManxPower: I'm in Huntsville now`
`04:10.59`	`rudholm`	`or phone lines`
`04:11.05`	`ManxPower`	`Qwell: Ah.`
`04:11.13`	`Strom_C`	`rudholm: what's disappointing is that the restaurant is now called "Gladstone's of Malibu"`
`04:11.23`	`Strom_C`	`which is stupid, because wasn't Malibu always GLobe 6?`
`04:11.24`	*** join/#asterisk ez` (n=ez@c66.110.149-45.clta.globetrotter.net)
`04:11.30`	`ManxPower`	`When a get a car (soon) I'll be about 2 hrs away from Digium`
`04:12.27`	`*** join/#asterisk bkruse_home (n=kruz@69.73.127.92)`
`04:12.29`	`ManxPower`	`I can't wait for spring/summer`
`04:12.30`	`Qwell`	`Sokol really needs to have Astricon in LA again this year`
`04:13.28`	`Strom_C`	`i think it is in LA this year`
`04:13.32`	`rudholm`	`I always thought Topanga was GLadstone5`
`04:13.39`	`Qwell`	`blitzrage said maybe not :(`
`04:13.44`	`Strom_C`	`rudholm: yeah, that's topanga`
`04:13.52`	`Strom_C`	`rudholm: I'm talking about malibu`
`04:14.05`	`rudholm`	`yeah, well, Gladstone's is in neither :)`
`04:14.05`	`Strom_C`	`wtf, qwell`
`04:14.08`	`rudholm`	`it's in Los Angeles :)`
`04:14.15`	`Strom_C`	`hehe`
`04:14.17`	`rudholm`	`yeah, wtf?`
`04:14.17`	`Strom_C`	`yup`
`04:14.23`	`rudholm`	`I'm not going if I have to go to AL`
`04:14.30`	`rudholm`	`how much does it cost, btw?`
`04:14.35`	`rudholm`	`I wonder if I can expense it.`
`04:14.37`	`Strom_C`	`rudholm: but we can eat at waffle house`
`04:14.39`	`Qwell`	`~$600?`
`04:14.47`	`rudholm`	`dang`
`04:14.51`	`rudholm`	`what do I get for that?`
`04:14.57`	`Qwell`	`dunno`
`04:15.05`	`rudholm`	`I paid less than that for LISA`
`04:15.13`	`*** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net)`
`04:15.15`	`rudholm`	`(well, I didn't pay anything at all)`
`04:16.02`	`rudholm`	`unfortunately, since I don't work in our voice messaging/voip group, I probably can't justify 600$ on an asterisk conf`
`04:16.40`	`Qwell`	`tell them google is sending somebody from their <insert your department> department`
`04:17.08`	`rudholm`	`haha`
`04:17.13`	`*** join/#asterisk ars247 (n=no@ftw-66-42-87-41.customer.stknca.fire2wire.com)`
`04:17.40`	`Qwell`	`what do you even do there? :p`
`04:17.42`	`rudholm`	`"But Google is sending someone from their jerk-off and do nothing department!"`
`04:17.47`	`Qwell`	`gotcha`
`04:17.53`	`Strom_C`	`qwell: he shows off his Telstar phone`
`04:18.05`	`rudholm`	`yes, and my white tone dial Ericofon`
`04:18.35`	`rudholm`	`and soon my 2C2 :)`
`04:18.45`	`rudholm`	`it's so darn heavy though`
`04:18.52`	`Strom_C`	`oh cool, so you are putting it on your desk at work`
`04:18.54`	`rudholm`	`gonna need a dolly to get it up here to my office`
`04:18.56`	`rudholm`	`yeah`
`04:19.02`	`rudholm`	`since I'll have three of them`
`04:19.17`	`rudholm`	`well, I'll have 2 2C2s and one 2D2`
`04:19.25`	`Strom_C`	`details, details`
`04:19.28`	`rudholm`	`yeah`
`04:19.34`	`*** join/#asterisk ars247 (n=no@ftw-66-42-87-41.customer.stknca.fire2wire.com)`
`04:19.47`	`rudholm`	`it'll snuggle right into the corner of my cubicle`
`04:20.05`	`rudholm`	`would be sweet if I could get the coin relay to work first, though`
`04:20.24`	`Strom_C`	`DO NOT PUSH`
`04:20.27`	`rudholm`	`hahahaah`
`04:20.30`	`rudholm`	`everyone does`
`04:20.37`	`rudholm`	`it's like the warning on silica gel`
`04:20.37`	`Strom_C`	`yep`
`04:20.40`	`rudholm`	`everyone eats it`
`04:20.48`	`rudholm`	`you have silica gel??`
`04:20.50`	`Strom_C`	`ive never eaten it, actually`
`04:20.52`	`rudholm`	`oh good, I'm hungry!`
`04:20.58`	`rudholm`	`haha, neither have I.`
`04:21.11`	`Strom_C`	`i tried pouring it into the toilet once`
`04:21.28`	`rudholm`	`and?`
`04:21.35`	`rudholm`	`I use it as cat litter now`
`04:21.38`	`rudholm`	`it's pretty good at that`
`04:22.14`	`*** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-144-90.ny325.east.verizon.net)`
`04:22.24`	`Strom_C`	`oh, cool`
`04:22.38`	`Strom_C`	`yeah, nothing exciting happened when i poured it in the toilet`
`04:25.11`	`*** join/#asterisk coppice (n=chatzill@13.168.17.210.dyn.pacific.net.hk)`
`04:26.37`	`*** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn)`
`04:28.23`	`*** join/#asterisk intralanman (n=lanman@pool-71-253-242-197.nrflva.east.verizon.net)`
`04:29.11`	`*** join/#asterisk mrc3_ (n=mrc3@189.157.107.61)`
`04:29.38`	`mrc3_`	`hello! anyone here with a pap2?`
`04:30.29`	`Strom_C`	`I've got one within kicking distance`
`04:31.17`	`mrc3_`	`mine is requesting the tftp file all right, i've got dns spoofed, but the http request never comes`
`04:31.32`	`mrc3_`	`in fact, there's never a dns query for httpconfig`
`04:31.44`	`AJaymn`	`;)`
`04:31.55`	`*** join/#asterisk AJaymn (n=boiwonde@24-159-236-181.dhcp.mdsn.wi.charter.com)`
`04:32.22`	`*** join/#asterisk bkruse_home (n=kruz@69.73.127.92)`
`04:32.32`	`mrc3_`	`i've got a pap2 with 3.1.9(LSc). is it possible to unlock it with that firmware version?`
`04:32.52`	`AJaymn`	`mrc3_ has it been on the internet before?`
`04:33.26`	`*** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)`
`04:33.55`	`mrc3_`	`it might have been. i blocked traffic to vonage with iptables (it's working fine), but i think i plugged it right away when i received, before blocking traffic`
`04:34.06`	`Strom_C`	`ah`
`04:34.11`	`mrc3_`	`i was able to access the pap2's web server, but not anymore`
`04:34.26`	`Strom_C`	`my key to unlocking mine was to create a network which was not physically connected to any other network`
`04:34.30`	`mrc3_`	`i set the user password, i could dial **** and all, but not anymore`
`04:34.45`	`*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)`
`04:35.24`	`mrc3_`	`my network setup seems to be fine, because it is effectively blocking network traffic to vonage's network`
`04:35.32`	`mrc3_`	`the thing is that my pap2 is not asking for the firmware`
`04:35.45`	`mrc3_`	`it doesn't look for it in the httpconfig.vonage.net server`
`04:36.23`	`mrc3_`	`i'm using tcpdump at all times`
`04:37.23`	`*** part/#asterisk bkruse_home (n=kruz@69.73.127.92)`
`04:38.45`	`mrc3_`	(for the record, my iptables rule was: `iptables -A FORWARD -s 192.168.1.82 -j DROP`; works pretty well!)
`04:40.30`	`*** join/#asterisk lyroy (n=lyroy@bas1-montreal02-1096575772.dsl.bell.ca)`
`04:41.41`	`lyroy`	`Does someone know why with an Cisco ATA 186 an Asterisk my ATA always disconnect after a certain time, and then when an inconmig call comes it goes straight to the unavailable message...`
`04:42.12`	`Strom_C`	`disconnects after a call has been up for a while?`
`04:42.47`	`lyroy`	`when I'm not using my phone fow a while`
`04:43.02`	`lyroy`	`the ATA seems to goes down`
`04:43.08`	`Strom_C`	`let me guess`
`04:43.15`	`Strom_C`	`the ATA is behind a router`
`04:43.36`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`04:43.57`	`lyroy`	`yeah right`
`04:44.23`	`Strom_C`	`is that "correct" or is that you being sarcastic?`
`04:44.43`	`lyroy`	`no no it is not workin ;)`
`04:44.57`	`lyroy`	`sorry about that..`
`04:45.12`	`Strom_C`	`so, let me rephrase, and please answer "yes" or "no":`
`04:45.17`	`Strom_C`	`is the ATA behind a router?`
`04:45.33`	`lyroy`	`yes`
`04:45.42`	`Strom_C`	`ok, and the asterisk box is in front of the router?`
`04:46.06`	`*** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net)`
`04:46.22`	`lyroy`	`no the asterisk box is on a public ip outside`
`04:46.32`	`Strom_C`	`right, so it's in front of the router`
`04:46.44`	`Strom_C`	`what's happening is that the router is closing those ports`
`04:47.27`	`lyroy`	`ok so is there a way to fix it?`
`04:47.32`	`Strom_C`	`either (1) shorten the registration interval, (2) have the ATA send keepalives, or (3) use qualify=yes in sip.conf`
`04:47.54`	`lyroy`	`wich one is the best`
`04:48.14`	`Strom_C`	`probably 1 or 2`
`04:48.22`	`matt_`	`hello, in asterisk.conf what is the line that points to the sounds dir ?`
`04:48.31`	`matt_`	`i have installed the sounds but asterisk isn't finding them`
`04:48.46`	`*** join/#asterisk deb_user (n=none@70-59-111-238.albq.qwest.net)`
`04:48.49`	`lyroy`	`In the Cisco ATA 186 the registration interval would be: SIPRegInterval??`
`04:48.59`	`deb_user`	`anybody out there using dundi? care to chat about its uses and implications?`
`04:49.06`	`matt_`	`humm, ok i have found them but they are in a different place`
`04:49.20`	`Strom_C`	`lyroy: probably`
`04:49.49`	`lyroy`	`hmm I alredy drop it from 3600 to 5`
`04:49.58`	`lyroy`	`and no success`
`04:50.03`	`Strom_C`	`5 seconds is a little too short`
`04:50.06`	`Strom_C`	`try 60 seconds`
`04:50.26`	`Strom_C`	`and then reboot the AT`
`04:50.28`	`Strom_C`	`er, ATA`
`04:51.17`	`lyroy`	`alright ill try it`
`04:56.55`	`matt_`	`can somebody help me please, i'm tring to use wakeup.php but it attemps to call asoon as i set it`
`05:00.11`	`*** join/#asterisk InHisName (n=Administ@c-68-38-105-1.hsd1.pa.comcast.net)`
`05:03.35`	`InHisName`	`how does testfeature work ?`
`05:05.34`	`lyroy`	`thx i think it will fix the problem`
`05:16.52`	`InHisName`	`am I the only human left listening ?`
`05:17.50`	`mrc3_`	`InHisName, nope. i'm human and i'm listening, but i can't really help`
`05:18.43`	`InHisName`	`mrc3_ can't help or just haven't learn much since (yesterday) when you started ?`
`05:19.09`	`Fr0zen_`	`Does asterisk work behind nat connecting to another sip server or stun server?`
`05:19.53`	`InHisName`	`As far as I understand works all ways. Not so sure as all at once though.`
`05:20.21`	`mrc3_`	`InHisName, haven't learnt about testfeature, can't provide insight`
`05:21.01`	`InHisName`	`I know too little also. Tried to blindly use it but nope, no luck there.`
`05:23.20`	`InHisName`	`Any running with Stanaphone, freedigits, icall.com, fone4life, onesuite, sunrocket ? Last 2 work fine for me, others I need some help with.`
`05:23.24`	`*** join/#asterisk d42 (n=don@124.189.39.16)`
`05:23.43`	`InHisName`	`HiYa d42`
`05:26.04`	`d42`	`I am attempting to setup 000 emergency dialing in Australia. The info found so far suggests calling 141162000982 from NSW VOIP number. However my VOIP Provider doesn't allow that number. Is there another prefix or number I can call?`
`05:30.06`	`InHisName`	`Here is USA, some have called the emergency no from neighbors house and asked how to call using standard numbering. Several got yelled at for not being an emergency. etc. Others called the NON emergency no and they sometimes were helpful.`
`05:33.50`	`InHisName`	`yawn sure is quiet tonight (morning)`
`05:34.42`	`*** join/#asterisk \|dennis\| (n=dennis@200.32.233.82)`
`05:34.58`	`InHisName`	`welcome back dennis`
`05:35.10`	`*** join/#asterisk cburn123 (n=chburnet@adsl-75-44-10-189.dsl.milwwi.sbcglobal.net)`
`05:35.19`	`cburn123`	`Good evening all.`
`05:35.35`	`InHisName`	`evening cburn123`
`05:36.24`	`cburn123`	`I am having a bit of an issue and man is it getting late!`
`05:37.16`	`InHisName`	`Late ? its only 39 minutes into the new day here! So what is the issue ?`
`05:39.51`	`cburn123`	`I installed the custom context module from aussievoip.`
`05:40.22`	`cburn123`	`After a reboot a problem cropped up that I'm not to sure how to fix..`
`05:41.51`	`cburn123`	`When I dial in on PSTN to my VegaStream Gateway it is route onto my lan as a SIP message. The Asterisk box stopped responding to my inbound route.`
`05:43.24`	`InHisName`	`cburn123 can you remove the custom context and problem goes away ?`
`05:43.29`	`cburn123`	`It's as if asterisk doesn't 1. get did or 2. doesn't respect the route`
`05:43.45`	`*** join/#asterisk sahafeez (n=sahafeez@ip68-6-215-70.sd.sd.cox.net)`
`05:43.47`	`InHisName`	`What does the custom context supposed to do?`
`05:44.02`	`cburn123`	`I have tried although I suspect unsuccesfully..`
`05:45.43`	`cburn123`	`What would be an easy way to direct my boss to this room.. he is having troubles getting here`
`05:45.47`	`cburn123`	`?`
`05:47.07`	`InHisName`	`A. needs a login`
`05:47.07`	`cburn123`	`he has that.. he has hjoined the asterisk channel vbut says he is the only one there`
`05:47.07`	`InHisName`	`B. needs to be validated. i.e. needs to have a password and be unique user name`
`05:47.44`	`cburn123`	`ahh wrong server..`
`05:47.47`	`InHisName`	`Only one there ? Never have seen less than 200. Must be spelling it wrong. /join #asterisk is way to do it.`
`05:48.05`	`InHisName`	`I am on freenode`
`05:48.41`	`cburn123`	`Same here.`
`05:48.46`	`cburn123`	`A-well..`
`05:48.51`	`cburn123`	`back to the problem..`
`05:49.37`	`cburn123`	`is there anyway for me to see spacifically how a call is being handled? I mean I have my debugs cranked up and this call only shows a few things.. None of which is informative.`
`05:50.06`	`InHisName`	`set verbose is high ?`
`05:50.30`	`cburn123`	`May I paste a section of the debug in here?`
`05:50.50`	`Qwell`	`~pb`
`05:50.54`	`jbot`	`methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste`
`05:50.56`	`InHisName`	`use one of the pastebin places .`
`05:51.28`	`cburn123`	`it's only about 4 lines.. that to much?`
`05:51.50`	`Qwell`	`yes`
`05:51.55`	`cburn123`	`k`
`05:51.55`	`InHisName`	`Other debug tools I have used is NoOp() put stuff in the () and it show on cli as it is tripped over while running.`
`05:52.14`	`cburn123`	`how do i enable that?`
`05:53.32`	`InHisName`	`exten => s,23,NoOp(user=${EXTEN}) and other nifty things as they change around.`
`05:55.05`	`cburn123`	`http://pastebin.ca/370065`
`05:56.04`	`InHisName`	`cannot find extension context 'Lane'`
`05:56.13`	`InHisName`	`what is 'Lane' ?`
`05:56.43`	`[TK]D-Fender`	`InHisName : A context you mentioned somwhere else in your config`
`05:57.16`	`cburn123`	`It's one of the "custom contexts" we made using the custom context module..`
`05:57.42`	`cburn123`	`This call comes in on a sip trunk and should be directed to an IVR`
`05:57.54`	`cburn123`	`somehow though the call is being pushed to ext 201..`
`05:58.01`	`InHisName`	`Understood. aparantly it is not where * is looking for it.`
`05:58.36`	`InHisName`	`what is the context name where the IVR is at ?`
`05:59.18`	`InHisName`	`what context does the sip trunk start with when a call comes in ?`
`05:59.43`	`cburn123`	`the IVR was created using freepbx and before any of this custom businees was installed. so it would be in the default for IVR`
`06:00.03`	`InHisName`	`the name of it is.....`
`06:00.06`	`cburn123`	`hehe`
`06:00.27`	`*** join/#asterisk elmerbug (n=don@dsl017-061-162.sfo4.dsl.speakeasy.net)`
`06:00.37`	`*** part/#asterisk mrc3_ (n=mrc3@189.157.107.61)`
`06:00.40`	`cburn123`	`Welcome ElmerBug`
`06:02.23`	`cburn123`	`context=from-trunk`
`06:02.41`	`cburn123`	`for the incomming settings on the sip-trunk`
`06:03.06`	`InHisName`	`So, cburn123 the IVR is in 'hehe' context ? and the start up is 'from-trunk' ?`
`06:03.29`	`*** join/#asterisk joelsolanki (i=joelsola@202.160.161.94)`
`06:03.33`	`cburn123`	`no hehe was me laughing.. I am looking for the IVR's context`
`06:03.42`	`InHisName`	`oh`
`06:03.45`	`joelsolanki`	`Good evening`
`06:04.07`	`*** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)`
`06:04.09`	`InHisName`	`I thought you were having fun with the 'Whos on first game'`
`06:04.16`	`cburn123`	`Are the IVRs normally kept in extensions.conf?`
`06:04.26`	`InHisName`	`yes`
`06:04.26`	`[TK]D-Fender`	`InHisName : No, Who's on second....`
`06:04.30`	`cburn123`	`heh`
`06:04.57`	`InHisName`	`whats on third ? I forgot why`
`06:05.03`	`[TK]D-Fender`	`cburn123 : Everything that gets processed on any call is in there.`
`06:05.40`	`sbingner`	`who's on first`
`06:06.07`	`InHisName`	`I have IVRs in 'house-day', 'house-night', 'office-day', and 'office-night'. I start with 'default'`
`06:06.26`	`sbingner`	`what's on second, and I Don't know's on third`
`06:06.28`	`kaldemar`	`[TK]D-Fender: unless other files are included in extensions.conf. ;)`
`06:06.52`	`[TK]D-Fender`	`kaldemar : Same thing.`
`06:07.25`	`joelsolanki`	`202.202.202.202 1419564207 708df1592ad 00102/00000 unkn No Tx: ACK`
`06:07.25`	`joelsolanki`	`202.202.202.202 1803796301 2b1c32e10ff 00102/00000 unkn No Tx: ACK`
`06:07.31`	`InHisName`	`search on #include to find the includes`
`06:07.32`	`cburn123`	`Hrmm I don't see any of my IVR's in extensions.conf`
`06:07.41`	`[TK]D-Fender`	`InHisName : It is a terrible idea to have a context so generically named as [default]. LIke what the hell does that imply as far as access goes? Bad idea, period.`
`06:07.43`	`joelsolanki`	`i see this when i do sip show channels`
`06:07.53`	`joelsolanki`	`and what i see this this calls are not connected.`
`06:08.15`	`InHisName`	`That is the way it came with install, I just left it that way.`
`06:08.18`	`[TK]D-Fender`	`cburn123 : the joys of running FreePBX or some other GUI. You get to use * and not have a CLUE where anything is or how it works.`
`06:09.16`	`cburn123`	`The joys of running it is figuring these things out, I mean I've got to start somewhere right?`
`06:09.28`	`InHisName`	`any you new guys try out testfeatures ? I have a question or two.`
`06:09.45`	`InHisName`	`yup, sure due, cburn123`
`06:09.49`	`[TK]D-Fender`	`InHisName : "came with install". Frankly * doesn't come with ANYTHING, and only if you install the samples do you even get the first pile of crap, for which you should only read and build your own from scratch. Sample files are bloated garbage without a proper sense of heirarchy, and forget the GUI created stuff entirely.`
`06:10.42`	`[TK]D-Fender`	`cburn123 : No. the GUI won't teach you anything. its bloated crap thats so damned complex its like trying to learn astrophysics before basket-weaving.`
`06:10.55`	`joelsolanki`	`?`
`06:11.00`	`joelsolanki`	`any hints`
`06:11.01`	`joelsolanki`	`202.202.202.202 1419564207 708df1592ad 00102/00000 unkn No Tx: ACK`
`06:11.01`	`joelsolanki`	`202.202.202.202 1803796301 2b1c32e10ff 00102/00000 unkn No Tx: ACK`
`06:11.14`	`InHisName`	`ptui samples are great, it now is 90% different than what I started with. I keep learning. name wasn't all that important to fix up.`
`06:11.16`	`[TK]D-Fender`	`cburn123 : Loaded full of crappy macro's populating all sorts of DB values without a proper understanding of WHY. That is no way to learn`
`06:11.16`	`joelsolanki`	`is this calls still connected ?`
`06:11.31`	`[TK]D-Fender`	`joelsolanki : "show channels". See anything there?`
`06:11.46`	`[TK]D-Fender`	`joelsolanki : I seriously doubt it`
`06:12.01`	`joelsolanki`	`show channels i dont see anything.`
`06:12.19`	`joelsolanki`	`means those calls are not there.`
`06:12.23`	`cburn123`	`So you suggest then that I ditch my production server and leave my company without a phone system while I go learn the proper way? I am not quite sure where you are going with this..`
`06:12.26`	`joelsolanki`	`but it is there in sip show channels`
`06:12.27`	`InHisName`	`DB values, hmmm, musta loaded the dummies version as I didn't get the deluxe samples after all.`
`06:12.36`	`joelsolanki`	`is this problem ?`
`06:12.59`	`*** join/#asterisk Phel (n=chatzill@adsl-156-209-188.mia.bellsouth.net)`
`06:13.14`	`[TK]D-Fender`	`cburn123 : See you are having to look at this backwards because you thought you could get somewhere at the start and work your way back. That was the mistake.`
`06:13.49`	`[TK]D-Fender`	`cburn123 : You'd have been better served to have waited a bit first, and take the time to learn how it works before dragging them into a pidgeon-holed setup.`
`06:13.56`	`Phel`	`Where would be a good place to ask general SIP questions. I cannot register to any SIP VSP to save my life`
`06:14.28`	`cburn123`	`No actually it was working great till I installed that module ..`
`06:14.30`	`[TK]D-Fender`	`InHisName : was referring to GUI generated configs. the "samples" that you can install along with * upon compile should be examined, but never used.`
`06:14.45`	`[TK]D-Fender`	`cburn123 : Which "module"?`
`06:14.48`	`elmerbug`	`Folks, I need to understand how to definitively determine what context asterisk has chosen on an inbound call?`
`06:14.51`	`InHisName`	`cburn123, try to undo that install then.`
`06:14.55`	`cburn123`	`custom context`
`06:15.00`	`elmerbug`	`Any guidance you can offer?`
`06:15.04`	`[TK]D-Fender`	`Phel : Specifically from *?]`
`06:15.05`	`cburn123`	`I needed multi tenant`
`06:15.24`	`Phel`	`[TK]D-Fender: What?`
`06:15.24`	`joelsolanki`	`<[TK]D-Fender> : ??? is this problem`
`06:15.35`	`[TK]D-Fender`	`cburn123 : Welcome to the "dead-end" of your system.`
`06:15.36`	`cburn123`	`I have already tried and aussievoip is down so I can consult the docs..`
`06:15.39`	`Phel`	`I can't do SIP VoIP`
`06:15.57`	`Phel`	`I've tried different service providers`
`06:16.03`	`[TK]D-Fender`	`joelsolanki : No, its probably just a lingering message where the other side shut up before * knew they acknowledged that its over.`
`06:16.07`	`elmerbug`	`Phel, please describe your calling scenario.`
`06:16.08`	`Phel`	`Can't register to anyone`
`06:16.23`	`[TK]D-Fender`	`Phel : What kind of errors do you get?`
`06:16.41`	`joelsolanki`	`hmm ok`
`06:17.28`	`Phel`	`Depends on what I do, but for example, in Ekiga, if I try to use a stun server (which I probably should since I am behind a NAT), it says it's blocked`
`06:18.02`	`Phel`	`Ekiga opens ports 5060-5063 and 1720, all of which I forward UDP`
`06:18.12`	`Phel`	`but no dice`
`06:18.12`	`*** join/#asterisk irq (n=dan@wsip-70-167-112-5.sd.sd.cox.net)`
`06:19.22`	`[TK]D-Fender`	`Phel : And I'm betting you have not done ANY of the sip.conf NAT settings required for your system to function.`
`06:19.40`	`[TK]D-Fender`	`Phel : And you need a heck of a lot more ports forwarded to *,e tc...`
`06:19.46`	`cburn123`	`If * was looking for a context it couldn't find and I can't get my IVR to play when I dial in could I create that context and have it play the IVR for a bandaid?`
`06:20.13`	`Phel`	`[TK]D-Fender: My nat is on the router`
`06:20.28`	`Phel`	`And I've even tried forwarding all prots`
`06:20.44`	`[TK]D-Fender`	`cburn123 : You need to actually provide the PROPER and expected context names.`
`06:21.03`	`InHisName`	`cburn123, you could, but not sure if that will help in long run.`
`06:21.06`	`[TK]D-Fender`	`Phel : There are a number of seetings you need to do under [general] in sip.conf`
`06:21.32`	`cburn123`	`how do you ignore someone?`
`06:21.37`	`InHisName`	`cburn123 need to find what 'Lane' is and rebuild it or make sure it is in right path.`
`06:21.43`	`Phel`	`I'm not using asterisk, just a softfone`
`06:21.52`	`Phel`	`cburn123: /ignore`
`06:21.57`	`cburn123`	`thanx`
`06:22.40`	`Phel`	`WHich is why I first asked where general sip questions would be appropriate`
`06:23.29`	`[TK]D-Fender`	`Phel : Ok, where are you trying to connect to?`
`06:23.58`	`Phel`	`sipphone for example`
`06:24.20`	`Qwell`	`cburn123: who you ignoring?`
`06:24.40`	`cburn123`	`Not important`
`06:24.50`	`[TK]D-Fender`	`Phel : Well if you're jsuta client behind NAT connecting to an outside service, and are onlya single VoIP device doing so, you generally don't need to forward anything.`
`06:24.52`	`Qwell`	`I would very highly recommend listening to [TK]D-Fender`
`06:25.12`	`cburn123`	`I do not have time to be lectured.. I only have time to move this issue forward.`
`06:25.22`	`Qwell`	`~hafc`
`06:25.24`	`jbot`	`somebody said hafc was hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out.`
`06:25.58`	`cburn123`	`Pulling my hair out is why I get up in the morning.`
`06:26.12`	`Phel`	`[TK]D-Fender: Hmm. I dunno. What's stun for then?`
`06:26.28`	`[TK]D-Fender`	`Phel : usually the other side will have a keep-alive which will allow you to work. Perhaps your ISP is blocking SIP. Many do this, especially if they offer VoIP services of their own or are in choots with another company that does`
`06:27.22`	`Phel`	`I am using a phone company ...`
`06:27.56`	`Phel`	`How could I really tell?`
`06:28.00`	`[TK]D-Fender`	`Qwell : Yeah some people shut down at the ealrier signs of resistance. Of course some people ignore anything but what they want to hear so, c'est la vie...`
`06:28.30`	`[TK]D-Fender`	`Qwell : I wasn't going to make anything of it. He just wants his answer so to that I say...`
`06:28.33`	`[TK]D-Fender`	`~wglwat`
`06:28.35`	`jbot`	`i heard wglwat is well, good luck with all that`
`06:28.53`	`[TK]D-Fender`	`Phel : I'm not entirely sure...`
`06:29.16`	`cburn123`	`Dude listening to you tell me how to learn is not benificial at the moment. It may make you feel superior and great and all that.. Great.. But I've got a problem to fix..`
`06:29.37`	`InHisName`	`wow, one I don't know, what is whlwat ?`
`06:29.56`	`Phel`	`Is there any way around such port blockage?`
`06:29.58`	`[TK]D-Fender`	`Phel : have you tried hooking Ekiga up to say FWD?`
`06:30.22`	`Phel`	`Would that be any different?`
`06:30.30`	`Phel`	`I'll try that if you think it would`
`06:30.33`	`[TK]D-Fender`	`Phel : What router BTW? Some are particularly bad with NAT and can really just KILL SIP. Some D-Links are like that, and Cisco PIX is a nightmare.`
`06:30.43`	`cburn123`	`Sorry InHisName.`
`06:30.55`	`[TK]D-Fender`	`Phel : Just another common service you could test with for free`
`06:31.27`	`[TK]D-Fender`	`InHisName : its exactly what jbot said.`
`06:31.52`	`InHisName`	`oh, yea`
`06:32.07`	`Phel`	`[TK]D-Fender: the router and Nat are built into my ISP's device`
`06:32.29`	`Phel`	`So it's a very weird setup`
`06:32.34`	`[TK]D-Fender`	`Phel : STUN is jsut a little trick to help your client know what kind of NAT its behind so it can try picking the best way to signal the other side toe nsure the best odds of traversal possible. It doesn't actually "FIX" anything per-se`
`06:32.43`	`*** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com)`
`06:32.54`	`[TK]D-Fender`	`Phel : Oh, so you're trying to get Ekiga running BEHIND that?`
`06:33.07`	`Phel`	`yes`
`06:33.48`	`Phel`	`My PC <-> NAT Router <-> Internet`
`06:34.17`	`[TK]D-Fender`	`Phel : Ah well thats a key factor. With any luck thats exactly the problem. It could very well be intercepting all SIP traffic and mangling everything up.`
`06:34.39`	`[TK]D-Fender`	`Phel : try another router`
`06:34.45`	`Phel`	`I don't have any other option :(`
`06:35.16`	`Phel`	`I suppose I could ask them to give me one without the router built in`
`06:35.39`	`[TK]D-Fender`	`Phel : Or just buy one yourself. Or temporarily jsut connect your internet connection to a PC.`
`06:35.43`	`[TK]D-Fender`	`(directly)`
`06:35.57`	`[TK]D-Fender`	`Phel : Just a temporary measure for the test`
`06:36.08`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
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`06:36.32`	`Phel`	`Naw, the DSL device and router are the same thing. So I have to get a plain DSL device`
`06:36.41`	`IguanaNed`	`Hello all`
`06:36.42`	`Phel`	`and a router`
`06:37.06`	`[TK]D-Fender`	`Phel : Oh wow... a SUPER all-in-one box... yeah tahts a bad combo and you are completely owned by it...`
`06:37.12`	`Phel`	`Yep`
`06:37.19`	`[TK]D-Fender`	`Phel : thats the exact sort of dependency you should avoid.`
`06:37.23`	`IguanaNed`	`anyone use a2billing for billing voip instead of calling cards`
`06:37.23`	`Phel`	`It's sooooo convenient`
`06:37.37`	`Phel`	`>:-(`
`06:37.47`	`cburn123`	`So if I have any DID/any CID as an inbound route. How could an inbound sip call with a 404 not found?`
`06:38.02`	`[TK]D-Fender`	`Phel : For the guy who wants the exacty service it was sold for use with perhaps... and the second thats not good enough, you're up a creek.`
`06:38.30`	`Phel`	`I was being facetious`
`06:38.48`	`[TK]D-Fender`	`Phel : I was being kind :)`
`06:38.50`	`Phel`	`I think I'm gonna curl up into a ball and sob now`
`06:39.19`	`[TK]D-Fender`	`Phel : No, now is a golden opportunity to rethink your infrastructure and plan how YOU want thing to work.`
`06:39.47`	`Phel`	`Under port forwarding, it has a "SIP Client" option`
`06:39.59`	`Phel`	`Which you'd think would help`
`06:41.22`	`[TK]D-Fender`	`Phel : Smart = dumb with most of these units....`
`06:42.00`	`IguanaNed`	`can anyone suggest a good program for billing voip`
`06:42.40`	`[TK]D-Fender`	`IguanaNed : best bet, check the WIKI, then the mailing lists (biz)`
`06:42.54`	`Phel`	`And also, after I did start forwarding ports, my PC seemed to make more progress. By that I mean, with nothing forwarded, it only opened 5060, but with 5060 forwarded, it opened 5060-5063 and 1720`
`06:43.15`	`[TK]D-Fender`	`Phel : like I said, you shouldn't even HAVE to forward anything....`
`06:43.48`	`Phel`	`Does anyone here run something I could point my client at. I'd like to get good info on what the server sees`
`06:44.06`	`[TK]D-Fender`	`Phel : Try FWD`
`06:44.08`	`Phel`	`And all the VOIP providers have sucky support`
`06:44.23`	`Phel`	`OKay`
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`06:53.22`	`AJaymn`	`Anyone know of a wholeseller that offers set cost for per outbound US calling trunk? (not per min) ?`
`06:54.43`	`SwK`	`none that are worth a damn`
`06:54.58`	`[TK]D-Fender`	`AJaymn : that doesn't sould like a "wholesaler" kind of product.`
`06:55.21`	`SwK`	`actually gerbil crossing has a service liket hat but I think its TDM only hand off`
`06:55.26`	`AJaymn`	`:(`
`06:55.27`	`[TK]D-Fender`	`AJaymn : individual unlimited accounts sure, but not massively multi-channel typically.`
`06:56.00`	`SwK`	`gerbil crossings it like 100USD/channel (+/- 10USD or so)`
`06:56.46`	`SwK`	`but depending on what you are doing and how many minutes you need, that can be had for far less`
`06:57.11`	`SwK`	`ie; 0.01/minute blended (maybe even better)`
`06:58.01`	`Phel`	`[TK]D-Fender: Registration Failed`
`06:58.38`	`[TK]D-Fender`	`Phel : code #?`
`06:59.33`	`Phel`	`Ekiga doesn't say. Question though. Is your account password = FWD password because FWD Number is not the same as User Name`
`06:59.45`	`Phel`	`And I am using FWD Number`
`07:01.00`	`[TK]D-Fender`	`Phel : I'm not sure on the precise settings, but I'd bet there's an Ekiga how-to on their site`
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`07:11.39`	`Phel`	`[TK]D-Fender: Yo, so I managed to get some debugging output and it keeps on saying "STUN could not create socket!"`
`07:11.55`	`*** part/#asterisk r0d3nt (n=RatMan@punk.valuetel.net)`
`07:12.07`	`Phel`	`And it's timeing out during registration`
`07:12.17`	`[TK]D-Fender`	`:/ who;s stun server are you pointing it at, and have you tried without it altogether?`
`07:12.31`	`JT`	`stun isn't necessary most of the time`
`07:12.47`	`JT`	`only for old retarded sip proxies/B2BUAs that were dumb`
`07:12.48`	`Phel`	`stun.fwdnet.net`
`07:13.04`	`[TK]D-Fender`	`JT : non * scenario. We're covering our bases here...`
`07:13.13`	`[TK]D-Fender`	`phel : ok, and completely without...?`
`07:13.20`	`Phel`	`trying now`
`07:13.39`	`JT`	`[TK]D-Fender: heh yeah, as i said, only shit servers need stun these days :)`
`07:14.44`	`Phel`	`Still times out but no more stun messages`
`07:14.59`	`[TK]D-Fender`	`Phel : debateable improvement ;)`
`07:15.26`	`[TK]D-Fender`	`Phel : Ok, you serioulsy need to test without that gateway of yours..... I am seriously distrusting it ATM....`
`07:15.38`	`Phel`	`"Set state Terminated_Timeout for transaction 15 REGISTER"`
`07:16.00`	`[TK]D-Fender`	`Phel : do you ahve a stop in Ekiga to TELL it your WAN IP?`
`07:16.20`	`[TK]D-Fender`	`Phel : if so plug it in. Typicaly its when the other side doesn't know where to respond to.`
`07:16.53`	`Phel`	`I assume that was a spot`
`07:17.37`	`Phel`	`No`
`07:18.02`	`Phel`	`But I could use siproxd as an outbound proxy right?`
`07:18.40`	`Phel`	`I know there's an option for that with it`
`07:21.33`	`[TK]D-Fender`	`Phel : I am pretty sure at this point its your gateways fault.....`
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`07:37.47`	`[TK]D-Fender`	`Ok, very late here... off to bed. later all, and GL`
`07:38.20`	`Phel`	`thanks`
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`08:36.24`	`brea`	`Anyone having issues with Global Crossing right now?`
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`09:18.19`	`foobar778`	`I have an fx0 port How to make another user be able to use this line for outgoing extensions.conf suggestion please`
`09:19.41`	`mosty`	`use the Dial command with a zap channel type`
`09:20.05`	`foobar778`	`mosty I dont have a zaptel device`
`09:20.16`	`foobar778`	`do I need one?`
`09:20.24`	`mosty`	`what FXO device do you have then?`
`09:20.43`	`foobar778`	`Im using a DVG-1120s router`
`09:20.54`	`foobar778`	`2fxs and one fx0`
`09:21.13`	`mosty`	`erm, does that thing run asterisk?`
`09:21.22`	`foobar778`	`yes`
`09:21.51`	`mosty`	`what driver does it use for the FXO port?`
`09:22.05`	`foobar778`	`same fxs phone is receiving pstn calls`
`09:22.25`	`foobar778`	`driver??`
`09:22.43`	`mosty`	`at the asterisk console, do "zap show channels"`
`09:22.50`	`foobar778`	`ok`
`09:23.42`	`foobar778`	`only pseudo`
`09:24.29`	`mosty`	`so that d-link box runs linux+asterisk?`
`09:24.49`	`foobar778`	`its the ata`
`09:25.22`	`foobar778`	`and working on awith asterix on debian`
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`09:25.49`	`mosty`	`i'm confused- how is it connected to your debian box?`
`09:26.06`	`foobar778`	`its is not`
`09:26.34`	`mosty`	`then what does your debian box have to do with this?`
`09:26.36`	`foobar778`	`modem>>router>>>dvg-1120s`
`09:26.49`	`foobar778`	`debain from another port on router`
`09:27.08`	`foobar778`	`the debain runs asterix`
`09:27.13`	`mosty`	`ok, and you are trying to configure the dvg-1120s to do what?`
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`09:27.52`	`foobar778`	`I want the other users to access the fx0 phone on the dvg-1120 to make outbound pstn calls`
`09:28.55`	`foobar778`	`dvg-1120 registers as 6001 user and can do both I want the other users in sip.conf to have same ability`
`09:29.21`	`foobar778`	`6001 can use pbx and pstn`
`09:29.22`	`mosty`	`you mean you want asterisk sip users to be able to dial out via the dvg-1120s?`
`09:29.28`	`foobar778`	`yes`
`09:29.39`	`foobar778`	`using the fx0`
`09:30.30`	`mosty`	`you need to setup the dvg-1120s as a sip server, create a sip account for asterisk, then dial an extension via that sip account`
`09:32.15`	`foobar778`	`mostydvg-1120 is a user in sip.conf user 6001\`
`09:32.47`	`foobar778`	`But how to make it a server??`
`09:33.19`	`mosty`	`type=peer`
`09:33.27`	`foobar778`	`as of now from the dvg-1120s is setup as automatic call redirect`
`09:33.32`	`mosty`	`http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+type`
`09:33.40`	`foobar778`	`ahhh`
`09:33.46`	`foobar778`	`I think I got u`
`09:34.09`	`mosty`	`then you can Dial(SIP/6001/<pstn number here>)`
`09:34.10`	`foobar778`	`friend wont do it??`
`09:34.33`	`mosty`	`friend if you want to send and receive calls from the device`
`09:34.38`	`foobar778`	`yes`
`09:34.44`	`foobar778`	`its friend now`
`09:35.00`	`foobar778`	`let me try that from console`
`09:35.16`	`foobar778`	`DIAL/6001//pstn`
`09:36.26`	`mosty`	`http://www.voip-info.org/wiki/view/D-link+DVG-1120`
`09:38.16`	`foobar778`	`reda that`
`09:38.46`	`foobar778`	`mosty the problem in the above DIAL was pstn no context`
`09:39.39`	`foobar778`	`<PROTECTED>`
`09:39.50`	`foobar778`	`6001 is in context from-sip`
`09:40.17`	`foobar778`	`anyway to redo DiIAL command syntax`
`09:40.55`	`foobar778`	`888 is set as prefix for redirection to pstn in dvg-1120s`
`09:42.43`	`foobar778`	`so I have several users 6001 which the dvg-1120s registers as 6002 .6003.6004`
`09:43.44`	`foobar778`	`so far when I pick up the analog phone and dail the prefix888 I have outbound pstn`
`09:44.20`	`foobar778`	`and if I get an incoming pstn the analog phone rings`
`09:44.57`	`foobar778`	`the same analog phone can act as fxs and dial all other extensions like 6002 as well`
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`09:45.57`	`foobar778`	`So the snag is to get 6002 to use the pstn line I dont even lnow if it is possible`
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`09:58.58`	`Aces1Up`	`has anyone here used astbill?`
`10:00.43`	`Aces1Up`	`does anyone here have any billing solutions for asterisk other than a2billing?`
`10:05.01`	`d42`	`My Austalian VOIP Provider does not allow 1411 (the Telstra prefix) calls in Australia. I want to set up 000 emergency calls, and have found documented that calls can be made to 141162000982 from VOIP phone in NSW.`
`10:05.29`	`d42`	`Does anyone know of another number that can be called to contact the 000 emergency call center?`
`10:05.35`	`JT`	`what provider is that?`
`10:05.48`	`d42`	`JT, it is Freecall`
`10:06.14`	`JT`	`hmm, never heard of them`
`10:06.42`	`d42`	`JT, they are freecall.net.au`
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`10:09.22`	`d42`	`Perhaps there may be another way of directing a call through the Telsra network?`
`10:09.57`	`JT`	`hardware :)`
`10:10.26`	`d42`	`JT, I'm thinking of a phone number that may be called?`
`10:11.27`	`JT`	`what's the point? if the voip provider doesn't support 000 then likely they don't send unique ANI with IPND details for you, so if you call 000 it won't give them sufficient information`
`10:13.01`	`d42`	`JT, The point is that in Australia, Government regulation provides interum service that designates VOIP location as nomadic by state, or nomadic nationally.`
`10:13.25`	`d42`	`JT, thus the 000 operator is prompted to ask for location.`
`10:13.31`	`JT`	`i'm in australia`
`10:13.42`	`JT`	`well it sounds like it's an issue with your voip provider`
`10:13.53`	`JT`	`if they can't help you, time to change provider`
`10:14.05`	`JT`	`looking at their plans, they're not very good value for money anyway`
`10:14.28`	`CrazyTux`	`Hey guys, does anyone have any good examples of asterisk and click to call, agi type stuff?`
`10:14.29`	`d42`	`JT, what is it about their plans that concerns you.`
`10:14.51`	`JT`	`price`
`10:15.03`	`d42`	`JT, can you be more specific?`
`10:15.04`	`JT`	`lack of features (as we can see here with 000)`
`10:15.18`	`JT`	`monthly fees are too high for the level of service`
`10:16.13`	`d42`	`JT, what would you recommend?`
`10:16.19`	`*** join/#asterisk flying_Luck (n=melifaro@ppp85-141-155-160.pppoe.mtu-net.ru)`
`10:16.44`	`JT`	`to be honest, i find engin not that bad considering how compliant their number plan is with a real telco`
`10:17.04`	`JT`	`they populate ipnd etc too, so they do proper telco stuff`
`10:18.17`	`d42`	`JT, you compain about the monthly fees. engin is twice the monthly fee, and for last month I saved $7 on local calls compared to what it would have cost if I were with engin.`
`10:19.06`	`JT`	`if you make tonnes of local calls that are less than 10mins, then, yes, you could save a couple of dollars`
`10:19.15`	`JT`	`umm if i'm reading freecall'`
`10:19.24`	`flying_Luck`	`Hello everybody, i'm trying to connect asterisk with nec neax 2000 station via E1 card. physical level seems to be ok, but i've got repeating <-- SABME -> UA from station without any RR. I'm using asterisk 1.2.13 with libpri 1.4.0. Where should i dig ?`
`10:19.27`	`JT`	`umm if i'm reading freecall's site right it's $5 + $7 for a real phone number`
`10:19.39`	`JT`	`$12`
`10:19.57`	`JT`	`it's $9.95 or so for the same with engin`
`10:20.12`	`JT`	`let's not forget a proper numbering plan and 000 service is worth something too`
`10:20.53`	`d42`	`JT, yes when you include the DID fee. However it is still an overall saving when compared to engin.`
`10:21.21`	`d42`	`JT, I think that the 000 issue is something I will resolve.`
`10:21.43`	`JT`	`well to be honest, $5 a month for no DID and outgoing only is ridiculous`
`10:22.00`	`JT`	`there's plenty of voip provders with no monthly fee for outgoing only in australia`
`10:22.27`	`d42`	`JT, How is a cheaper plan than engin rediculous? Why aren`
`10:22.49`	`d42`	`JT, Why are you not using one with no monthly fee?`
`10:22.49`	`JT`	`it is not cheaper for the same level of functionality, we've been through this`
`10:23.08`	`JT`	`because i want a real phone number and better service`
`10:23.15`	`JT`	`i am actually using some with no monthly fee too`
`10:23.23`	`JT`	`i use a number of service providers`
`10:24.06`	`d42`	`JT, which ones do you use, and why?`
`10:24.24`	`JT`	`i'm also sceptical whether freecall would support callerid sending, to be honest`
`10:24.31`	`JT`	`if you had a did`
`10:25.09`	`JT`	`antratel is an interesting one, cheapest calls to mobiles for no monthly fee, and you can choose your routes for calls, for price vs quality`
`10:25.17`	`JT`	`within australia and overseas`
`10:25.21`	`JT`	`very interesting system`
`10:25.28`	`JT`	`strictly outgoing only, no CID sending`
`10:26.29`	`JT`	`i tried pennytel but their system seemed like a bit of a joke, i wasn't able to recharge to attempt calls that cost`
`10:26.35`	`JT`	`might give it another go some time`
`10:26.45`	`JT`	`as quality was good for free numbers, and their paid prices are ok`
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`10:31.08`	`JT`	`ok as in cheap, not as cheap as antratel`
`10:31.34`	`JT`	`but yeah if you want something close to a proper phone line service, only the more expensive/polished providers offer that`
`10:31.50`	`redax`	`hi`
`10:31.54`	`JT`	`and engin has good mobile call rates on their biz 50 plan, which is good for business`
`10:32.13`	`redax`	`can't compile zaptel-1.2.10 on newer kernels...`
`10:32.56`	`Aces1Up`	`quick question, what is the password if i use this line to reset my password on my mysql database?`
`10:32.56`	`Aces1Up`	`SET PASSWORD FOR root@localhost=PASSWORD('my_new_password');`
`10:33.42`	`redax`	`compile warning like: kmem_cache_t is deprecated (declared at include/linux/slab.h:17)`
`10:34.25`	`redax`	should be the `my_new_password' ;-)
`10:35.41`	`Aces1Up`	`man, thats not workin :(`
`10:35.46`	`d42`	`JT, The reason I chose freecall is they provide IAX support, and an overall cheeper deal than engin. Engine does not provide support if you want to connect up a Trixbox.`
`10:35.48`	`redax`	although I use `mysqladmin -p password my_new_password
`10:36.57`	`redax`	`what kernel would you use with a bristuffed asterisk (bristuff-0.3.0-PRE-1y)`
`10:37.46`	`JT`	`d42: well i guess that's not an issue with me, i don't need or want provider technical support on configuration, only with issues or faults with their service`
`10:37.57`	`Aces1Up`	`crap, now i locked myself out of mysql, is there anyway to reset my password?`
`10:38.12`	`JT`	`freecall is definately not cheaper than engin on the biz 50 plan for calls to mobiles if you make a lot of them a month`
`10:38.24`	`JT`	`20c/min`
`10:38.24`	`d42`	`JT, I hope you don't have any faults with their service. Your on your own.`
`10:38.35`	`JT`	`d42: err no, they support faults`
`10:38.43`	`JT`	`not typing in sip settings`
`10:39.20`	`JT`	`also they have one of the longer hours support lines of the .au voip providers`
`10:39.25`	`redax`	`have a nice weekend`
`10:40.01`	`JT`	`and yeah, trixbox, i'll reserve what i think of it :)`
`10:40.24`	`d42`	`JT, they kind of do and then they don't. Your using non approved equipment. So unless you can clearly demonstrate that the problem is there end, your in trouble. Anyhow had some very dismissive experiences with them.`
`10:41.02`	`JT`	`sure, but i can clearly tell if it is their end or not, so i guess it's been a non-issue so far`
`10:41.27`	`JT`	`it's not premium support, but not much is in the voip world`
`10:41.43`	`JT`	`margins are very low`
`10:42.28`	`d42`	`JT, you appear to be very one eyed. Do you represent engin?`
`10:42.39`	`Aces1Up`	`anyone know how i reset my mysql database server password?`
`10:42.56`	`JT`	`no, i'm hardly one eyed, i pretty much gave a run down of the disadvantages as well as advantages`
`10:43.32`	`JT`	`my only ongoing complaint with engin is they have RTP silence supression turned on coming from their end and they won't turn it off`
`10:43.45`	`foobar778`	`Jt do u use fx0 ports??`
`10:44.00`	`JT`	`but that's a minor quality issue when you compare it to most other voip providers' quality`
`10:44.58`	`JT`	`also, as a rule, i don't trust voip providers to be reliable, but engin has proved fairly reliable`
`10:44.58`	`foobar778`	`JT: do u use or are u familair with using fx0 ports`
`10:45.08`	`JT`	`foobar778: it's "FXO" not "fx0"`
`10:45.19`	`JT`	`Foreign eXchane Office`
`10:45.24`	`JT`	`eXchange`
`10:45.32`	`flying_Luck`	`Aces1Up, http://dev.mysql.com/doc/refman/5.0/en/resetting-permissions.html`
`10:45.34`	`foobar778`	`ok wyes I know do u use them`
`10:45.41`	`JT`	`foobar778: not really`
`10:45.51`	`JT`	`i have some gear here that does FXO`
`10:46.00`	`JT`	`but yeah, i try to use digital where possible`
`10:46.07`	`foobar778`	`my problem is I want other sip users to access the pstn line`
`10:46.18`	`JT`	`right, that should be do-able`
`10:46.27`	`foobar778`	`to me JT?`
`10:46.34`	`JT`	`yes`
`10:46.45`	`foobar778`	`I need a little help here`
`10:47.04`	`JT`	`what is the problem that you are experiencing?`
`10:47.06`	`foobar778`	`where do u want me to begin`
`10:47.24`	`foobar778`	`my setup?`
`10:47.30`	`foobar778`	`perhaps`
`10:47.56`	`foobar778`	`ok here is my setup`
`10:48.03`	`JT`	`well what's the problem, you can't work out how to do it?`
`10:48.55`	`foobar778`	`modem>>router>>dvg-1120s 2fxs and one fx0 with automatic call redirection`
`10:49.06`	`JT`	`d42: btw i don't think engin walk on water, none of the providers do, they just offer a good cost/features proposition, i've found`
`10:49.25`	`foobar778`	`can call out using phones in fxs ports to pstn`
`10:49.39`	`JT`	`hrm`
`10:49.41`	`foobar778`	`but other sip users cant`
`10:49.48`	`JT`	`well i'm not sure how this dvg-1120s works`
`10:49.59`	`JT`	`what's auto call redirection?`
`10:50.05`	`foobar778`	`I have a pps on it`
`10:50.22`	`foobar778`	`auto call allows a prefix to dial to pstn`
`10:50.32`	`JT`	`pps?`
`10:50.49`	`foobar778`	`888 will hit the pstn even though default line is voip`
`10:51.12`	`foobar778`	`powerpoint slideshow on 1120s is pps`
`10:51.22`	`foobar778`	`ppt maybe`
`10:53.58`	`foobar778`	`Jt: http://afterburn.no-ip.info:8050/160.jpg`
`10:54.44`	`d42`	`JT, I think you work for engin. I can tollerate no CID on outgoing. Other than issues with 000 emergency, which I will resolve, there just are cheeper plans than engin. So I think you over dramatise the differences, and different people value different things. Respect the differences, and don't push your view.`
`10:55.20`	`foobar778`	`JT link works now`
`10:55.30`	`foobar778`	`<PROTECTED>`
`10:55.32`	`JT`	`lol work for engin`
`10:55.55`	`JT`	`you're a little paranoid d42`
`10:56.39`	`*** join/#asterisk af_ (n=getsmart@ip-202-133.sn2.eutelia.it)`
`10:56.44`	`foobar778`	`JT: did u see the picture??`
`10:56.51`	`JT`	`not yet`
`10:57.22`	`foobar778`	`http://afterburn.no-ip.info:8050/160.jpg`
`10:57.22`	`JT`	`d42: does everyone conspire against you?`
`10:57.26`	`JT`	`YES YOUVE PASTED IT THREE TIMES foobar778`
`10:57.28`	`JT`	`argh`
`10:57.43`	`foobar778`	`ok`
`11:00.12`	`JT`	`d42: i know different people value different things, which is clearly why i've qualified all my statements with reasons and particular advantages or disadvantages`
`11:00.24`	`foobar778`	`Hey it migt be in the routing table`
`11:00.27`	`JT`	`i'm happy to move from engin the moment a superior deal comes out`
`11:00.30`	`JT`	`foobar778: most likely`
`11:00.48`	`foobar778`	`I will take a creenshot of it`
`11:00.54`	`foobar778`	`screenshot`
`11:01.15`	`d42`	`JT, what you read as paranoia was intended as humor. Sorry is seems it didn't come across as intended. My intention here is not to say my way is the best way or anything like that.`
`11:01.40`	`JT`	`d42: okay`
`11:01.46`	`d42`	`JT, I'm just after some information.`
`11:02.24`	`JT`	`right, well i don't think it's easy to properly call 000 without it being supported`
`11:02.40`	`JT`	`sure there are some dodgy pabxes like parliment house's that'll let you do it`
`11:02.46`	`JT`	`but probably not advisable to us`
`11:02.47`	`JT`	`e`
`11:02.47`	`*** join/#asterisk hackeron (n=hackeron@gentoo/user/hackeron)`
`11:02.52`	`*** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br)`
`11:03.05`	`foobar778`	`http://afterburn.no-ip.info:8050/800.jpg JT what do u think`
`11:03.18`	`hackeron`	`hey, I'm looking for a test number in the UK that's always engaged - anyone know where I can find one?`
`11:03.27`	`d42`	`Does anyone know of an alternate phone number that may be called to access the telsta network beside the 1411 prefix?`
`11:03.49`	`JT`	`foobar778: yes it's relevant, you really need to read up on the dlink documentation`
`11:04.23`	`JT`	`d42: most people here aren't in australia`
`11:04.53`	`foobar778`	`I have more documentation on that model than dlink has and what I have is minimal`
`11:05.40`	`foobar778`	`I called dlink they use whats on the web I found documentation elsewhere`
`11:06.08`	`JT`	`yeah dlink sucks`
`11:06.15`	`JT`	`i guess you may need to use trial and error`
`11:07.36`	`foobar778`	`yes JT I have another shot coming and when u see it Ill tell u what Im thinking for ur feedback`
`11:08.31`	`foobar778`	`http://afterburn.no-ip.info:8050/8001.jpg`
`11:09.40`	`foobar778`	`so if I have other sip users`
`11:09.59`	`foobar778`	`looks like I can define a number`
`11:10.33`	`foobar778`	`and it will route the user to the pstn`
`11:11.08`	`foobar778`	`so say user is 6006 and asterisk is 192.168.1.113`
`11:11.27`	`foobar778`	`then those will go in the entries`
`11:11.57`	`foobar778`	`and just define a number and it will route is that ur gues JT?`
`11:13.31`	`JT`	`yeah probably`
`11:13.35`	`JT`	`i've never used the unit`
`11:13.40`	`JT`	`so not sure`
`11:13.53`	`d42`	`I wish to enable calls to the 000 emergency service in Australia. The documentation says that calls may be made to 141162000982 from VOIP phones in NSW.`
`11:13.56`	`d42`	`My VOIP Provider does not allow 1411 calls. Is there an alternate phone number that may be called to access the Telstra network, or of calling the 000 emergency call center?`
`11:14.47`	`JT`	`d42: what documentation says this?`
`11:15.37`	`d42`	`JT, If you don't know the answer, please leave it clear for someone else to answer.`
`11:16.21`	`JT`	`i am asking a genuine question, please don't tell me how to act in this channel again`
`11:16.26`	`JT`	`anyone is free to respond`
`11:16.30`	`JT`	`i am hardly blocking them`
`11:16.35`	`JT`	`i'm curious too`
`11:19.04`	`JT`	`d42: if you check the australian numbering plan, there may be an alternative number there`
`11:32.04`	`*** join/#asterisk usn (n=usn@p54932F11.dip0.t-ipconnect.de)`
`11:32.40`	`usn`	`hi there - I am desperately looking for a documentation for capicommand - is there a listing of available options?`
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`11:37.05`	`*** join/#asterisk Mike800 (n=mike800@ip68-231-212-10.oc.oc.cox.net)`
`11:37.26`	`Strom_C`	`hi`
`11:37.32`	`Mike800`	`hello :p`
`11:37.57`	`Strom_C`	`whatcha doin up this late?`
`11:38.05`	`Mike800`	`need some help...`
`11:38.13`	`Mike800`	`with the application map in features.conf`
`11:38.18`	`Strom_C`	`lay it on me`
`11:38.33`	`*** part/#asterisk s1gny\|wrk (n=s1gny@p549177E6.dip.t-dialin.net)`
`11:38.59`	`Mike800`	`basically, i want to create a key sequence that will transfer someone to a queue, or different voicemail boxes`
`11:39.26`	`Mike800`	`:-)`
`11:39.46`	`*** join/#asterisk stoffell (n=stoffell@d51A4D4C3.access.telenet.be)`
`11:39.48`	`Strom_C`	`flesh it out; i think i know how to do it`
`11:40.01`	`Mike800`	`tough one`
`11:40.14`	`Mike800`	`really?`
`11:40.54`	`Mike800`	`lemme re-explain what i want it to do`
`11:41.49`	`Strom_C`	`well, i just said "flesh it out"; i dont want you to re-tread :)`
`11:41.53`	`*** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it)`
`11:42.06`	`Mike800`	`i dont know what that means :-)`
`11:42.28`	`Strom_C`	`it means "please explain in greater detail"`
`11:42.46`	`Strom_C`	`or it means "dead hookers"`
`11:42.51`	`Strom_C`	`one of the two :)`
`11:42.56`	`Mike800`	`Key Sequence #1: Places call on hold, asks employee what extension to transfer to. Key Sequence #2: Transfers call to a specific queue`
`11:43.18`	`Strom_C`	`ah ok, so you're having a human transfer inbound calls?`
`11:43.21`	`Mike800`	`:-)`
`11:43.26`	`Mike800`	`ya`
`11:43.35`	`Mike800`	`not even necessarily inbound`
`11:43.47`	`Strom_C`	`what kind of telephone sets are you using?`
`11:43.49`	`Mike800`	`it could be an outbound call that needs to be transferred as well`
`11:43.52`	`Mike800`	`Linksys SPA-942`
`11:44.01`	`Strom_C`	`there should be a transfer key on the set`
`11:44.04`	`Mike800`	`(not enough customizable buttons for these things)`
`11:44.16`	`Mike800`	`ya, but the secretaries would rather just press #1 or something`
`11:44.35`	`Strom_C`	`eh, inband transfers are icky; what if they're calling an IVR menu that requires them to press #?`
`11:44.39`	`Mike800`	`right now they have to press transfer, extension, transfer aain`
`11:45.00`	`Mike800`	`well, voip-info says as long as its not #1, you're ok`
`11:45.37`	`Strom_C`	`well....they press #, then the next selection is 1. or 2. or 3. or....`
`11:46.04`	`Mike800`	`if they press it fast enough, then it will perform the application map function...otherwise, it will just go through`
`11:46.22`	`Strom_C`	`i suppose it's worth a go`
`11:46.52`	`Mike800`	`its asterisk 1.2.15`
`11:46.58`	`Mike800`	`just for reference`
`11:46.59`	`Mike800`	`:-)`
`11:48.03`	`Strom_C`	`why not just uncomment the blindxfer line, modify it to whatever you want it to be, and leave it at that?`
`11:48.25`	`Strom_C`	`i don't see any benefit to having an application mapping when you can just do blind transfers to extensions in your dial plan`
`11:49.27`	`Mike800`	`well, they seem to want it... :-\ they're lazy people....to do a blind transfer on that phone, they have to do 3 key sequnces (not including the keys they have to press for the transfer extension)...and they thing it will be easier this way`
`11:49.40`	`Strom_C`	`no no no`
`11:49.46`	`Strom_C`	`look in the config file`
`11:49.52`	`Strom_C`	`there's a featuremap, and there's an applicationmap`
`11:49.57`	`Mike800`	`ohh`
`11:50.06`	`Mike800`	`ya...but like, transferring to a specific queue?`
`11:50.32`	`Strom_C`	`yeah, leave #1 as blind xfer, and then have sales on 500, support on 501, billing on 502...etc`
`11:50.55`	`Mike800`	`(i wish my dads company was that simple....;p)`
`11:51.11`	`Strom_C`	`the last PBX I set up has seven queues`
`11:51.18`	`Strom_C`	`and four queue members`
`11:51.46`	`Mike800`	`my dads has 15-20`
`11:51.49`	`Mike800`	`:-)`
`11:52.01`	`Strom_C`	`20 queues?`
`11:52.03`	`Mike800`	`and about 10 members :-)`
`11:52.03`	`Mike800`	`ya`
`11:52.07`	`Strom_C`	`??`
`11:52.09`	`Strom_C`	`why 20 queues`
`11:52.19`	`Mike800`	`cause each person has their own personal queue`
`11:52.33`	`Mike800`	`that other employees can transfer people into`
`11:52.43`	`Mike800`	`it works well for their business..`
`11:52.53`	`Strom_C`	`well ok, this comes to mind:`
`11:53.02`	`*** join/#asterisk lorinc (n=ang@caracas-3853.adsl.interware.hu)`
`11:53.06`	`Strom_C`	`you have the 20 sets as, say, extensions 501-520, right?`
`11:53.14`	`Mike800`	`ok`
`11:53.27`	`Strom_C`	`map your queues such that 601 is the queue for 501, 602 is the queue for 502, etc etc etc`
`11:53.45`	`Strom_C`	`or, even better`
`11:53.52`	`Strom_C`	`make it so that /every/ call to the telephone set goes into a queue`
`11:54.00`	`Mike800`	`it does`
`11:54.04`	`Mike800`	`and i already have it set up that way`
`11:54.24`	`Strom_C`	`ah ok, so you're at least two steps ahead of me already :)`
`11:54.40`	`Mike800`	`my issue is that the secretaries would rather have a 2-digit number that completes the full transfer to a specific queue...`
`11:54.48`	`Strom_C`	`ahhh`
`11:54.49`	`Strom_C`	`ok`
`11:55.13`	`Mike800`	`(i made the mistake of giving tem my e-mail address...i get at least 2 e-mails a day from them asking me for this...i made the mistake of telling them this was possible)`
`11:55.45`	`Strom_C`	`so just # followed by a 2 digit code?`
`11:55.53`	`Mike800`	`ya...`
`11:56.22`	`Mike800`	`or it could be *#(number)`
`11:56.33`	`Strom_C`	`yeah, there you go`
`11:56.34`	`Mike800`	`ive never had to enter *# on any IVR`
`11:56.42`	`Strom_C`	`map *# to blind transfer`
`11:56.55`	`Strom_C`	`and then have a separate context with two-digit mappings to the queues`
`11:57.03`	`Strom_C`	`make sure they dont conflict with your existing numbering plan`
`11:57.09`	`Strom_C`	`wham, bam, thank you sir`
`11:57.11`	`Mike800`	`lol`
`11:57.17`	`Mike800`	`but thats not using the app map`
`11:57.18`	`Mike800`	`haha`
`11:57.32`	`Strom_C`	`it's going to be too much trickery to use the app map`
`11:57.40`	`Mike800`	`i guess`
`11:57.45`	`Mike800`	`i cant find any documentation on it`
`11:57.46`	`Strom_C`	`unless...`
`11:57.50`	`Strom_C`	`one moment`
`11:57.54`	`Mike800`	`nothing that really teaches me much`
`12:00.02`	`Strom_C`	`how about this`
`12:00.09`	`*** part/#asterisk Aces1Up (n=really@ip68-227-41-148.lv.lv.cox.net)`
`12:00.14`	`Strom_C`	`er, no`
`12:00.15`	`Strom_C`	`never mind`
`12:00.42`	`Mike800`	`hehe`
`12:01.51`	`Strom_C`	`yeah, applicationmap for this would be a huge kludge`
`12:02.01`	`Mike800`	`:p`
`12:02.06`	`Mike800`	`thats why its gonna be fun`
`12:02.20`	`Strom_C`	`just use blindxfer in conjunction with some dialplan jiggerypokery and you're set`
`12:02.48`	`Strom_C`	`unless you want to add an application mapping for each queue`
`12:03.00`	`Strom_C`	`so #21, #22, #23, #24, etc etc etc`
`12:03.57`	`Mike800`	`yup`
`12:04.00`	`Mike800`	`thats what i wanna do`
`12:04.10`	`Strom_C`	`so, yeah, just do:`
`12:04.31`	`Strom_C`	`er no`
`12:04.39`	`Strom_C`	`that will just queue the secretary`
`12:04.44`	`Mike800`	`hehe`
`12:04.44`	`Strom_C`	`wake up, strom`
`12:05.03`	`Mike800`	`i dont like the way they have it set up....its so confusing`
`12:06.02`	`Mike800`	`:p`
`12:06.21`	`Strom_C`	`holy cocks, it's 4 AM`
`12:06.29`	`Mike800`	`i know!!`
`12:06.29`	`Strom_C`	`brb, switching to laptop`
`12:06.33`	`Mike800`	`its ok`
`12:06.38`	`Mike800`	`we'll work on it tomorrow`
`12:06.44`	`Mike800`	`i didnt realize its this late`
`12:07.39`	`JT`	`2307 here :)`
`12:08.01`	`Mike800`	`where is 'here'?`
`12:10.16`	`JT`	`sydney, australia`
`12:10.23`	`Mike800`	`ah..`
`12:11.55`	`Strom_M`	`internets`
`12:12.02`	`Mike800`	`hehe`
`12:13.34`	`Mike800`	`hey...im gonna go sleep`
`12:13.41`	`Mike800`	`i cant keep my eyes open`
`12:13.51`	`Mike800`	`thanks for your help strom`
`12:14.26`	`Mike800`	`i'll talk to ya tomorrow`
`12:14.31`	`Strom_M`	`alright`
`12:14.35`	`Strom_M`	`sleep well :)`
`12:14.39`	`Mike800`	`thanks :)`
`12:14.42`	`Mike800`	`good night`
`12:17.19`	`Strom_M`	`((( IN STEREO )))`
`12:17.47`	`JT`	`omg stereo`
`12:18.42`	`macTijn`	`(where available)`
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`13:26.19`	`*** join/#asterisk public- (n=PubliC@cpe-65-29-113-80.twmi.res.rr.com)`
`13:27.20`	`public-`	`good morning`
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`13:46.21`	`usn`	`Hi folx. I want to perform all kinds of actions AFTER a dial command. I know that dial does not come back, if everything went good. But how can I work around that behaviour?`
`13:50.27`	`*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)`
`13:50.31`	`PakiPenguin`	`hi everyone`
`14:06.05`	`*** join/#asterisk Ebola (n=Ebola@host86-142-178-37.range86-142.btcentralplus.com)`
`14:10.19`	`*** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net)`
`14:12.40`	`ManxPower`	`usn: the "h" extenson will be executed when the caller hangs up`
`14:15.38`	`public-`	`anyone using chan_cellphone?`
`14:16.04`	`public-`	`actually.. 2 questions.. anyone using a 7940 in SIP mode and have the time displayed on the screen?`
`14:33.20`	`*** join/#asterisk apardo (n=apardo@87.217.145.137)`
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`14:53.32`	`Qwell`	`public-: I'm using chan_cellphone some`
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`14:58.41`	`public-`	`Qwell: any idea if it works with the call waiting on the cell phone?`
`14:58.48`	`Qwell`	`doubt it`
`14:58.53`	`public-`	`ok, didn't see anything`
`14:58.59`	`public-`	`was wondering if I missed somethign`
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`15:01.27`	`snickerdoodle`	`Is there a seperate channel for Asterisk 1.40 suppport? Or same? (Topic suggests I might be in the wrong place)`
`15:02.05`	`uwe`	`hello, im trying to write expect script to be used with cisco IP Phones (telnet) to simulate real load on asterisk, i thought ill ask if someone already did something similar? well, does anyone know of something similar?`
`15:02.05`	`Nugget`	`telnet is eeeeeeevil!`
`15:02.29`	`uwe`	`hmm... i really hope Nugget is a bot`
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`15:07.15`	`coppice`	`I suppose you want them to wait and just watch the TV series of minds`
`15:08.04`	`*** join/#asterisk adker (n=chatzill@74-33-198-79.br1.glv.ny.frontiernet.net)`
`15:11.38`	`*** join/#asterisk marty-ott-athome (n=me@host-209-50-87-76.dyn.295.ca)`
`15:12.51`	`marty-ott-athome`	`Good Morning! Been a few week since I've been here.. we're about to try ASterisk for the first time. I've got a caller setup - however, the only thing I'm missing is the routing of calls... and this is an easy one (probably in the handbook).. but`
`15:13.10`	`marty-ott-athome`	`could someone tell me how to send ALL calls to an IP address of another SIP server?`
`15:13.59`	`joaovianna`	`Anyone using asterisk with video ?`
`15:14.31`	`w9sh`	`Hi Marty. I'll try and help.`
`15:14.45`	`marty-ott-athome`	`Im not 100% on it but I think that video is not yet 100% define for SIP..`
`15:14.49`	`marty-ott-athome`	`w9sh.. cool!`
`15:15.05`	`w9sh`	`are u using trixbox, freepbx or all with a handedited dialplan`
`15:15.41`	`marty-ott-athome`	`handedited dialplan - is there an easier way? I thought Trixbox and freepbx were simply alternatives to Asterisk`
`15:16.00`	`cburn123`	`not alts.. just front-ends`
`15:16.05`	`w9sh`	`they are gui's around asterisk`
`15:16.05`	`joaovianna`	`marty-ott-athome: Thanks, I'm using a Grandstream 3000 but I could not put video to work yet.`
`15:16.17`	`Qwell`	`poor ones, at that`
`15:16.24`	`marty-ott-athome`	`oh... cool.. I'll use trixbox in the future then`
`15:16.28`	`marty-ott-athome`	`oh..`
`15:16.38`	`w9sh`	`so what do u want to do to start. a home system or at the office or what?`
`15:16.47`	`Qwell`	`marty-ott-athome: I wouldn't recommend that my grandmother use trixbox`
`15:17.18`	`marty-ott-athome`	`Well, I've got a Mediatrix gateway setup (Quell -ok - any suggestions) and my voip only calls work fine. I just need to send my calls out to my SIP upstream provider`
`15:17.19`	`w9sh`	`true qwell, but i would have my grandmother try and do anything with asterisk :)`
`15:17.47`	`marty-ott-athome`	`Maybe trixbox then would be good for an admin person who needs to add a customer ..`
`15:18.16`	`w9sh`	`marty what OS and cpu hardware?`
`15:18.39`	`marty-ott-athome`	`3 ghz - 1 gig ram - intel, asus, FreeBSD`
`15:18.42`	`snickerdoodle`	`Can someone suggest where I might get advice / support on T.38 passthrough on Ast 1.40 ???`
`15:18.50`	`w9sh`	`and do you have the box doing anything else that you care about?`
`15:19.24`	`marty-ott-athome`	`just ssh`
`15:19.24`	`Qwell`	`snickerdoodle: I have yet to see you ask a question about it`
`15:19.24`	`marty-ott-athome`	`this box is dedicated to Asterisk`
`15:19.35`	`w9sh`	`that's too nice a machine for *!`
`15:19.54`	`w9sh`	`got anything older laying around?`
`15:20.15`	`snickerdoodle`	`I have an Asterisk 1.40 install with H323 in one leg, and SIP out the other. This works great for audio (g729 codec), but attempts to passthrough T.38 cause a disconnect.`
`15:20.18`	`marty-ott-athome`	`which reminds me - I have to ( :) ) -- think we're doing G711 right now - I'll have to get license for G729 -- our upstream will do both so it's non issue`
`15:20.23`	`marty-ott-athome`	`lol!!`
`15:20.35`	`Qwell`	`snickerdoodle: I don't think h323 supports t.38 passthrough`
`15:20.51`	`snickerdoodle`	`For some reason Asterisk says the RTP channel is switching to ulaw (even though both legs are g729), and then hangups up because it can't transcode`
`15:21.21`	`Qwell`	`snickerdoodle: get the latest 1.4 branch from svn.. that g729/ulaw thing has been fixed`
`15:21.27`	`Qwell`	`at least, I think it has`
`15:21.49`	`w9sh`	`well i just got called to daddy taxi duty so i gotta jump out. marty send me an email to w9sh@arrl.net and i can call you and chat while i drive.`
`15:21.51`	`snickerdoodle`	`Do you mean in general H323 can't do T38 passthrough - or just Ast 1.40 ?`
`15:22.04`	`public-`	`Qwell: does chan_cellphone support any of the fax services?`
`15:22.19`	`Qwell`	`public-: no`
`15:22.34`	`marty-ott-athome`	`wow! I appreciate that but I'm sure I can figure it out here.. it's a one-line I'm sure in the Dialplan`
`15:22.37`	`Qwell`	`public-: it's just an audio gateway for now`
`15:22.47`	`public-`	`Qwell: ok, thanks. Just thinkin outside the box. :)`
`15:22.50`	`Qwell`	`or, rather, it's just a headset`
`15:22.51`	`marty-ott-athome`	`it's cool.. thx though`
`15:22.56`	`Qwell`	`it supports audio gateways`
`15:23.20`	`Qwell`	`I'm sitting on a patch right now to support headsets though :D`
`15:23.22`	`w9sh`	`well maybe 6 lines, k cu`
`15:23.27`	`Qwell`	`then sms is next, if t-mobile comes through`
`15:23.47`	`Qwell`	`public-: and I did briefly mention fax over bluetooth the other day`
`15:23.50`	`marty-ott-athome`	`ok.. anyone - I'm sure I can my answer in the handbook but its an easy questoin.. I simply need to send all (6 lines eh.. hmmm) my calls to my upstream sip provider in my dial plan ... sorta like`
`15:24.00`	`Qwell`	`public-: the response was "...why?"`
`15:24.23`	`marty-ott-athome`	`exten => * sip:209.50.x.x`
`15:24.32`	`marty-ott-athome`	`anyone?`
`15:24.41`	`public-`	`Qwell: my question being... why not? it makes sense to use the phone's capabilities... especially for people who lack phone lines. P:\|`
`15:24.54`	`Qwell`	`public-: it's new yet.. sure it's possible, but...yeah`
`15:25.11`	`public-`	`Qwell: agreed.. I was suprirsed when I saw chan_bluetooth lose support.`
`15:25.43`	`marty-ott-athome`	`anyone? send my calls to my upstream sip provider?`
`15:25.54`	`Qwell`	`well, I went out and bought like $60 of bluetooth gear so I could play with it :p`
`15:26.03`	`Qwell`	`public-: it'll probably eventually support more stuff`
`15:26.10`	`public-`	`Qwell: I've got a motorola v3 and one of their headsets`
`15:26.24`	`public-`	`don't think this phone has a BT fax profile though`
`15:28.31`	`Qwell`	`is the v3 called something else?`
`15:28.36`	`Qwell`	`or...oh, that's a razr, isn't it?`
`15:28.48`	`public-`	`yah`
`15:28.49`	`public-`	`razr`
`15:29.08`	`public-`	`tryint to find a good tftp server for gentoo`
`15:29.14`	`*** join/#asterisk darken_darken (n=marco@239.143.76.83.cust.bluewin.ch)`
`15:30.01`	`snickerdoodle`	`Qwell: I'm a little lost...if I have an H323-SIP channel up with RTP using g729, and then the caller wants to send a fax using T.38, it looks like the RTP channels should close and T.38-over-UDPTL should start. I see UDPTL support in the sip.conf and (according to Open H323 web site) in H.323. Why/where would this fail?`
`15:30.19`	`dlynes_laptop`	`public-: gentoo doesn't ship with hpa tftpd?`
`15:31.16`	`public-`	`I don't have it installed by default`
`15:31.19`	`public-`	`I did a source build`
`15:31.27`	`public-`	`so all of my packages are custom`
`15:31.51`	`dlynes_laptop`	`damn`
`15:31.55`	`dlynes_laptop`	`that would suck`
`15:32.00`	`dlynes_laptop`	`talk about giving your hard drive a workout`
`15:32.02`	`public-`	`it takes some time to install`
`15:32.02`	`public-`	`:)`
`15:32.11`	`dlynes_laptop`	`and you did that on purpose?`
`15:32.14`	`public-`	`yah`
`15:32.21`	`public-`	`higher customization`
`15:32.24`	`public-`	`don't end up with as much BS`
`15:32.30`	`public-`	`however... I think where i'm at now, I've got the BS anyways`
`15:32.31`	`public-`	`:)`
`15:32.52`	`dlynes_laptop`	`so you should've started with a stripped down version of the a and d series from slackware, instead`
`15:32.57`	`coppice`	`snickerdoodle: openh323 may have T.38 support, but I don't think anything has been done to ie that into the rest of *`
`15:33.00`	`dlynes_laptop`	`forget gentoo :0`
`15:33.44`	`snickerdoodle`	`The 1.40 docs/wiki suggest T.38 support is in now...at least on the SIP side.`
`15:33.54`	`Qwell`	`public-: I use net-ftp/tftp-hpa`
`15:33.55`	`dlynes_laptop`	`snickerdoodle: t.38 passthrough`
`15:34.08`	`public-`	`Qwell: I just installed atftp`
`15:34.12`	`public-`	`looks like this will do the job`
`15:34.16`	`dlynes_laptop`	`snickerdoodle: and i'm guessing it's probably not well tested`
`15:34.18`	`coppice`	`passthrough support is there for SIP, but no work has been done on chan_h323`
`15:34.20`	`public-`	`right now using a windows tftp client..`
`15:34.20`	`public-`	`:\|`
`15:34.22`	`public-`	`err`
`15:34.23`	`public-`	`server`
`15:35.17`	`snickerdoodle`	`hmmm....does that mean that T.38 will not work in my case (h323 one leg, SIP the other) ? I'm using open H.323 - does it replace chan_h323?`
`15:37.05`	`elmerbug`	`Good morning, folks!`
`15:39.08`	`*** join/#asterisk chad123 (n=chburnet@adsl-75-44-10-189.dsl.milwwi.sbcglobal.net)`
`15:39.12`	`chad123`	`Morning`
`15:39.16`	`elmerbug`	`I have a VERY interesting question`
`15:39.53`	`dlynes_laptop`	`snickerdoodle: depends on whether you're using openh323 as a sip<=>h323 gateway, or if you're just using it as an interface for one of the h323 channel drivers in asterisk`
`15:39.55`	`marty-ott-athome`	`Can nyone tell me how to send all my sip calls to an upstream sip provdier by IP i.e. (" _* sip:ip_address`
`15:40.10`	`elmerbug`	`Inbound SIP calls with DIDs that match an inbound route are being handled by * as internal calls (extension calls)`
`15:40.13`	`elmerbug`	`Why???`
`15:40.15`	`dlynes_laptop`	`marty-ott-athome: set up a sip peer`
`15:40.24`	`Qwell`	`elmerbug: because your dialplan says to`
`15:40.52`	`elmerbug`	`Qwell: ok, thanks. Can you suggest how to identify and kill the culprit?`
`15:40.53`	`Qwell`	`computers only do what you tell them to do`
`15:41.00`	`Qwell`	`split up your contexts`
`15:41.19`	`marty-ott-athome`	`dlynes... ok.. isn't as simple as exten => s,1,Dial(SIP/ipaddress)`
`15:41.21`	`marty-ott-athome`	`something like that?`
`15:41.42`	`elmerbug`	`Qwell: Can you expand that a bit? Thanks for assisting.`
`15:42.43`	`dlynes_laptop`	`marty-ott-athome: you could do that, too`
`15:42.51`	`snickerdoodle`	`dlynes_laptop: I'm not sure I follow. I'm trying to create a SIP-H323 gateway, have installed Open H323, and am letting Asterisk do it's thing to bridge the SIP and H323 legs. Asterisk should just handoff the T38 stuff from one leg to the other I thought...`
`15:42.54`	`*** join/#asterisk redax (n=redax@r6.hu)`
`15:42.55`	`*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)`
`15:43.03`	`dlynes_laptop`	`marty-ott-athome: assuming you don't need to authenticate`
`15:43.08`	`dlynes_laptop`	`marty-ott-athome: or register`
`15:43.19`	`elmerbug`	`Qwell: For example, it wuld help if I could run a debug detailed enough to actually catch it in the act.`
`15:43.24`	`marty-ott-athome`	`I don't`
`15:43.33`	`dlynes_laptop`	`snickerdoodle: are you doing sip to openh323, or are you using h323 to it?`
`15:43.35`	`marty-ott-athome`	`my upstream simply permits me though some access-list`
`15:43.40`	`dlynes_laptop`	`snickerdoodle: openh323 has a sip stack now`
`15:43.43`	`redax`	`can anyone help me to figure out why does segfaults asterisk (bristuffed) with an OpenVox B400P card?`
`15:43.52`	`marty-ott-athome`	`So... is that the right way to do it then?`
`15:43.55`	`redax`	`when chan_zap loaded it segfaults`
`15:44.19`	`dlynes_laptop`	`marty-ott-athome: yeah, that should work then, but you'll need a different format`
`15:44.35`	`elmerbug`	`Qwell?`
`15:44.36`	`dlynes_laptop`	`marty-ott-athome: check the docs on the wiki for sip dial format`
`15:44.50`	`snickerdoodle`	`dlynes_laptop: I'm reaching the limits of my knowledge here..so I'll answer as best I can. I installed Open H323 which I though replaces the built-in H323?`
`15:44.51`	`marty-ott-athome`	`or is it.. like: exten => N,1,Dial(SIP/ip) ... I amreading hte handbook righ tnow`
`15:44.56`	`marty-ott-athome`	`but I'm not sure...`
`15:45.09`	`marty-ott-athome`	`looking for a section on send my call to a sip peer.`
`15:45.26`	`dlynes_laptop`	`snickerdoodle: the built-in h323 (h323 channel driver from asterisk-addons) doesn't require openh323`
`15:45.37`	`dlynes_laptop`	`snickerdoodle: there's three or so other h323 channel drivers that do require openh323`
`15:46.14`	`dlynes_laptop`	`snickerdoodle: however, you can just set up a sip peer to route all your calls to openh323, also because the new openh323 has both a SIP stack and an H.323 stack`
`15:46.45`	`dlynes_laptop`	`snickerdoodle: the version of openh323 that supports both is called 'Opal'`
`15:47.54`	`elmerbug`	`Folks, a debugging "how to" question. How do I turn up debugging so that I can see how * is handling incoming SIP calls to a context?`
`15:47.55`	`snickerdoodle`	`dlynes_laptop: I'm sinking fast (getting lost). Should I remove my Open H323 insallation and use the H323 from asterisk addons?`
`15:48.05`	`*** join/#asterisk kanaeda (i=kanaeda@CPE-76-178-154-170.natnow.res.rr.com)`
`15:48.28`	`dlynes_laptop`	`snickerdoodle: if you want, but save yourself the headache, and avoid the h323 channel drivers in asterisk`
`15:48.34`	`dlynes_laptop`	`snickerdoodle: install opal instead`
`15:49.34`	`dlynes_laptop`	`snickerdoodle: http://www.openh323.org/opal.html`
`15:49.39`	`snickerdoodle`	`dlynes_laptop: I'm going to do some quick reading on OPAL. BRB. Thanks`
`15:49.42`	`*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)`
`15:51.17`	`dlynes_laptop`	`snickerdoodle: It's up to version 2.0.0 now`
`15:51.25`	`dlynes_laptop`	`snickerdoodle: so, that document isn't quite up to date`
`15:51.51`	`dlynes_laptop`	`snickerdoodle: it's part of openh323 v2.0.0`
`15:51.59`	`dlynes_laptop`	`snickerdoodle: donm't need to grab it from svn/cvs anymore`
`15:52.43`	`dlynes_laptop`	`snickerdoodle: you can also do a search on the main openh323 homepage for the term 'opal'`
`16:03.53`	`Ryushin`	`How close do the zaptel version have to be to the released asterisk code?`
`16:04.36`	`Ryushin`	`I'm trying to use asterisk-1.2.15 with zaptel 1.2.12. I'm getting a compile error for asterisk.`
`16:08.09`	`Nugget`	`"a compile error" is not very helpful.`
`16:08.28`	`Ryushin`	`Nugget: Okay, let me give you the error. :)`
`16:09.06`	`snickerdoodle`	`Still a bit lost. Anyone...is OPAL a replacement to the latest OpenH323 ?`
`16:10.08`	`snickerdoodle`	`openh323.org offers Oh323 v1.12 but cannot download OPAL. Sourceforge offers Oh323 v1.19 and OPAL v.2.2.`
`16:11.24`	`Ryushin`	`Nugget: http://www.pastebin.ca/370457`
`16:11.48`	`Ryushin`	`It looks like 1.2.14 compiled fine.`
`16:12.33`	`Ryushin`	`I'm having problems compiling wanpipe with anything newer than zaptel-1.2.14 so I have to use zaptel-1.2.12.`
`16:12.54`	`*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)`
`16:16.20`	`Nugget`	`Hrm, that's not very helpful to me either. :) Hopefully someone else will have a better idea.`
`16:16.23`	`Nugget`	`Strange error.`
`16:17.24`	`Nugget`	`http://bugs.digium.com/view.php?id=8727 may be related.`
`16:21.11`	`Ryushin`	`thanks Nugget, I'll look at the bug.`
`16:22.32`	`*** join/#asterisk Pepse (n=pepse@ip68-109-169-37.ph.ph.cox.net)`
`16:29.23`	`grinsbalu`	`re`
`16:29.46`	`grinsbalu`	`one of you got the sip firmware for cisco 7941 and can send it?`
`16:40.15`	`uwe`	`um, can anyone help with how to know what the status of cisco 7940 phone is via telnet? is the speaker just open, or is there actually a call taking place!`
`16:44.19`	`*** join/#asterisk sergeus (i=guru-dev@195.112.98.13)`
`16:45.53`	`sergeus`	`Can anybody share Tormenta2 (T400P) manual with me?`
`16:46.59`	`coppice`	`the only real documentation is at www.zapatatelephony.org`
`16:48.03`	`*** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no)`
`16:49.19`	`sergeus`	`coppice: Thanks a lot! will dig that site`
`16:51.10`	`sergeus`	`coppice: there is a radial switch on my T400P with positions 0-f, can you explain me /briefly/ what is it?`
`16:53.11`	`JerJer`	`sergeus: i believe that is a card identifier`
`16:54.09`	`Qwell`	`JerJer: device state stuff is in trunk ;)`
`16:54.28`	`JerJer`	`Qwell: i saw that - kick ass stuff`
`16:54.47`	`JerJer`	`i'm gonna have to bust out my 7910 and 12SP here soon`
`16:55.06`	`Qwell`	`haven't tested it on a 7910 yet... no more room on my desk`
`16:55.11`	`*** join/#asterisk truescot (n=jaja@g192216.upc-g.chello.nl)`
`16:55.32`	`Qwell`	`should work fine though`
`16:55.57`	`JerJer`	`great - it will be a good thing for me to do then`
`16:56.39`	`truescot`	`can anyone suggest any good predictive dialers too me? i have installed and tested gnudialer, vicidial and sinedialer so far, any other i should be looking at?`
`16:57.34`	`truescot`	`i'm not a telemarketer :)`
`16:57.54`	`truescot`	`i am just playing and seeing whats possible`
`16:58.01`	`coppice`	`I predict that anyone receiving calls from a predictive dialer will be pissed off`
`16:58.02`	`Qwell`	`riiiight`
`16:58.10`	`truescot`	`i use asterisk for business and am just trying to learn everyting`
`16:58.33`	`sergeus`	`JerJer: so this switch should be engaged if i have a few cards in the same server, right? what if i have only 1 tor2 card? what should i set there? 0?`
`16:58.42`	`truescot`	`seriously i work for a private bank, we have no need for a dialer, but i am learning more and more on asterisk and would liek to look at dialers`
`16:58.52`	`Qwell`	`so that you can telemarket`
`16:58.57`	`truescot`	`:)`
`16:59.23`	`JerJer`	`truescot: write your own`
`16:59.33`	`JerJer`	`that will ensure good learning`
`16:59.33`	`truescot`	`not much need in that field but the private banking sector has to phone loads of clients every day to update them on their profile`
`16:59.43`	`truescot`	`good point,`
`16:59.55`	`coppice`	`and we all believe the tooth fairy will visit you when a pissed off call recipient knocks your teeth out :-)`
`16:59.57`	`JerJer`	`thank you, drive thru`
`17:00.07`	`truescot`	`:)`
`17:01.03`	`truescot`	`i work for www.oyens.nl not exactly a telemarketing company, when you are fishing for multi million euro accounts telemarketing doesnt tend to help ":)`
`17:05.00`	`coppice`	`dunno. I get telemarkers trying to sell me houses in other countries for investment.`
`17:06.15`	`truescot`	`does anyone actually buy from these assholes tho?`
`17:06.31`	`Qwell`	`of course not`
`17:06.35`	`truescot`	`i think they tend to go outta business quite quickly if thats ther only form of introduction`
`17:06.55`	`truescot`	`when they come in just place them on hold`
`17:07.07`	`truescot`	`see how long they are willing to wait`
`17:07.13`	`truescot`	`bait them a bit`
`17:08.19`	`coppice`	`that's what we've been trying to do with you`
`17:09.44`	`coppice`	`these are big property companies and banks calling me. they aren't scams, in that they are genuinely trying to sell property. they're just bloody annoying`
`17:09.57`	`truescot`	`my dad got pissed off with a double glazing company that kept calling, so on a day when he was bored called them and spent 4 hours going thru their range then told them the have nothing he wants so stop fin calling :`
`17:11.39`	`*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)`
`17:13.28`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`17:13.28`	`*** mode/#asterisk [+o mog] by ChanServ`
`17:13.34`	`Qwell`	`mog: !!!`
`17:14.23`	`sweeper`	`truescot: just say "take me off your list, or I will call bell and report you as an annoyance call"`
`17:14.43`	`Qwell`	`sweeper: I doubt companies in .nl care about bell :p`
`17:14.51`	`*** join/#asterisk sasch (n=sasch@host102-30-static.107-82-b.business.telecomitalia.it)`
`17:15.01`	`mog`	`Qwell, !`
`17:15.02`	`sweeper`	`Qwell: um, they care if they don't want bell to stop accepting calls from them :P`
`17:15.15`	`sweeper`	`unless you mean, he lives in nl,`
`17:15.39`	`sweeper`	`in which case, s/bell/${TECLO_PROVIDER_OF_YOUR_CHOICE}/`
`17:17.19`	`sasch`	`any person have a swissvoice ip10s ???`
`17:17.23`	`sergeus`	`found my misterious switch in Assembler's manual :)`
`17:17.26`	`sergeus`	`<PROTECTED>`
`17:18.34`	`sergeus`	`however, still no explanations about it's usage..`
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`17:23.33`	`sweeper`	`mmmm`
`17:23.55`	`sweeper`	`anyone have recommendations for call record web frontends?`
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`17:42.36`	`vlrk`	`hello all i have zapte card tdm400p in that i have 03 04 only with fxo`
`17:43.17`	`vlrk`	`the problem iam facing is when i configure in zaptel.conf as fxoks=3-4`
`17:43.46`	`vlrk`	`it says that fxs are configured with fxo signalling`
`17:43.50`	`vlrk`	`any ideas`
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`17:44.39`	`Cybertoy`	`hi .. anyone know if zfone works with asterisk?`
`17:45.13`	`Cybertoy`	`if I understand zfone correct you need to start it .. .and then start the phone client.. and zfone acts as "proxy" ... so in theory asterisk could be the client, no?`
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`17:51.13`	`Qwell`	`vlrk: swap your signalling`
`17:51.24`	`Qwell`	`vlrk: You are supposed to configure it with fxs signalling`
`17:52.22`	`flying_Luck`	`anybody can point me to a free q.921/q.931 software protocol analyzer ? asterisk intense debug seems not enough`
`17:53.06`	`kb1_kanobe`	`flying_Luck: Asterisk dumps the raw hex when you ask for 'pri debug span...', so you only need a copy of the relevant tables.`
`17:53.35`	`kb1_kanobe`	`A completel copy of q931 is hard to come by, but most of the used IEs are documented on the web`
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`17:53.46`	`kb1_kanobe`	`s/completel/complete`
`17:53.51`	`vlrk`	`Qwell: but what i have is fxo not fxs`
`17:53.59`	`Qwell`	`vlrk: yes, trust me, switch it`
`17:55.40`	`vlrk`	`i did that , it says configured but when i do ztcfg`
`17:55.51`	`vlrk`	`it shows as fxs that 03 and 04`
`17:56.15`	`Qwell`	`did you swap it in both configs?`
`17:56.50`	`vlrk`	`Qwell what do you mean that fxs to 3-4 and fxo to 1-2`
`17:56.55`	`vlrk`	`if that is then i did that`
`17:57.08`	`vlrk`	`and in ztcfg it shows as 4 channels configured`
`17:57.24`	`Qwell`	`in zaptel.conf and zapata.conf`
`17:57.29`	`vlrk`	`ok`
`17:58.17`	`sasch`	`anyone use swissvoice ip10 phone ??`
`17:58.21`	`vlrk`	`yes i did`
`17:58.33`	`kb1_kanobe`	`flying_Luck: I should clarify - asterisk dumps verbose decoding of the IE's when you issue 'pri debug span...' and also shows the raw hex in []. If it doesn't understand an IE then you'll get the raw hex and an explanation of what wasn't understood. Eg. http://bugs.digium.com/view.php?id=9058`
`17:58.51`	`sasch`	`<vlrk> do you use swissvoice phone ??`
`17:59.12`	`vlrk`	`Qwell it does not detect`
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`18:01.01`	`flying_Luck`	`kb1_kanobe, i've got itu-t q.921 pdf. My problem really is at q.921 or lower level. with 2 different software setups (same configs) i've got either cycling SABME/UA or just SABME from asterisk side without any UA from pbx side. I really don't know where should i dig`
`18:02.21`	`kb1_kanobe`	`flying_Luck: heh. you're on the right track already then, so you probably need to ask this question in asterisk-dev. It'll be burried somewhere in the libpri code.`
`18:02.50`	`kb1_kanobe`	`good luck :-)`
`18:03.03`	`flying_Luck`	`tnx :)`
`18:03.47`	`vlrk`	`sasch: what s that swissvoice phone ?`
`18:05.28`	`sasch`	`i have a swissvoice sip phone ... i configure it but when connect to my server`
`18:05.31`	`sasch`	`asterisk say`
`18:05.32`	`sasch`	`Saved useragent "Swissvoice IP10 SP v1.0.1 (Build 4) 3.0.5.1" for peer 15`
`18:05.39`	`vlrk`	`Qwell i pasted my configs here http://pastebin.ca/370551`
`18:05.47`	`sasch`	`but in my telephone is write connect to proxy server`
`18:06.10`	`vlrk`	`and i connected the phone line to 3 rd jack in card`
`18:06.28`	`vlrk`	`when i dial to that number it does not show any signals ..`
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`18:11.21`	`smellybelly`	`ug... I'm having some awful zaptel trouble... can anyone help?`
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`18:12.32`	`*** mode/#asterisk [+o anthm] by ChanServ`
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`18:33.30`	`DocHolliday`	`hey Qwell`
`18:33.38`	`Qwell`	`I'm not here.`
`18:33.54`	`DocHolliday`	`qwell the 7941 works :)`
`18:34.04`	`Qwell`	`with chan_skinny?`
`18:34.12`	`DocHolliday`	`SIP`
`18:35.03`	`Qwell`	`Then I don't much care :P`
`18:35.24`	`DocHolliday`	`skinny is more important to you?`
`18:35.29`	`Qwell`	`of course`
`18:35.42`	`DocHolliday`	`why? :O`
`18:36.05`	`Qwell`	`because it's a much nicer protocol`
`18:36.24`	`DocHolliday`	`heh okay..`
`18:43.25`	`*** join/#asterisk deb_user (n=Hypnotis@albuquerque.agroinnovations.com)`
`18:44.04`	`shido6`	`heh`
`18:44.33`	`deb_user`	`I have an fxo interfaces that rings an fxs interface right away on incoming calls, then after 20 sec. it rolls over to voicemail, only problem is the fxs still rings for a few seconds after it is already in voicemail, any dialplan suggestions to remedy this?`
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`18:46.24`	`ruied`	`(join #debian`
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`19:25.37`	`DeLmAr`	`Hi everyone. Soft-switch.org appears to be down and I am looking for a tarball of a recent version of spandsp 0.0.2 . Does anyone here have a copy I can get a hold of please?`
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`19:28.48`	`fetcher`	`I have spandsp-0.0.2pre20.tar.gz, but that's probably out of date by now (downloaded it in '05)`
`19:30.52`	`KnowWhat`	`but i want to know if there should be directory /var/run/mysql??`
`19:34.15`	`DeLmAr`	`fetcher, that might help tho`
`19:35.27`	`DeLmAr`	`fetcher, hrm. ok. my debian distro has 0.0.2pre26-1`
`19:36.36`	`DeLmAr`	`when i apply the patch to the makefile, and copy the rx/tx fax apps to the apps folder.. everything compiles as expected... yet when i load asterisk, it dislikes the fax modules.`
`19:37.00`	`tzafrir_home`	`DeLmAr, the .orig.tarball from a Debian mirror is just as good`
`19:37.15`	`DeLmAr`	`tzafrir, you would think. but i cant make it happy.`
`19:37.51`	`tzafrir_home`	`the patch to the apps dir is highly version dependent`
`19:38.09`	`tzafrir_home`	`You should be able to patch it manually yourself`
`19:38.16`	`DeLmAr`	`tzafrir, yeah.. and I only have a patch+apps from an earlier version...`
`19:38.23`	`DeLmAr`	`tzafrir, the patch works`
`19:38.32`	`Qwell`	`coughbug coppicecough :P`
`19:38.39`	`*** join/#asterisk Bobthehunter (n=Bobthehu@145-27.mc.cite.net)`
`19:38.41`	`tzafrir_home`	`the apps normally don't change between versions`
`19:39.04`	`DeLmAr`	`tzafrir, as i say.. i can patch it ok.. and it compiles without an issue but when i start asterisk, it has symbol issues with the two modules.`
`19:39.16`	`DeLmAr`	`asterisk 1.2.15`
`19:40.18`	`DeLmAr`	`so if anyone has a working spandsp + patch + rx/txfax apps combination that works with 1.2.15... id love to get ahold of them.. or even for an earlier version of Asterisk.`
`19:42.39`	`Carp1`	`if I do Background(file) someone should be able to enter an extension during the prompt, right?`
`19:44.02`	`DeLmAr`	`should do.`
`19:45.53`	`tzafrir_home`	`well, as qwell said, the maintainer of soft-switch.org is currently in the channel`
`19:49.09`	`DeLmAr`	`tzafrir, i have no idea who that is im afraid.... no offense :P .. but I don't spend alot of time in here`
`19:55.41`	`DeLmAr`	`perhaps the maintainer could give an idea as to when soft-switch.org might be back online? perhaps its online but its a 'route' issue between here and there... /shrug`
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`20:03.35`	`DeLmAr`	`hrm. is it possible to setup something with Asterisk & spandsp to provide remote access .. ie. a ppp session, from an extension on the IVR? kinda thinking about having an alternative means to access my system if my DSL goes down and needs intervention.`
`20:04.23`	`DeLmAr`	`almost always comes back up itself, but once a year it goes down and the modem is lazy.`
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`20:10.20`	`*** join/#asterisk n\|cotine (i=nicotine@147.202.49.52)`
`20:10.31`	`n\|cotine`	`Any nat gurus around?`
`20:10.45`	`IguanaNed`	`anyone use a2billing for billing voip instead of calling cards`
`20:13.12`	`*** part/#asterisk IguanaNed (n=you@CPE000625db3f84-CM00111ae43f1e.cpe.net.cable.rogers.com)`
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`20:14.15`	`DeLmAr`	`n\|cotine dunno about guru but im not bad at dealing with nat, altho the best way is to just avoid it :P`
`20:14.50`	`n\|cotine`	`DeLmAr: It just seems that placing our company desk phones on public IPs is not the best way to do things.`
`20:15.15`	`DeLmAr`	`n\|cotine what kinda phones?`
`20:15.33`	`n\|cotine`	`DeLmAr: Sipura/Linksys and Cisco.`
`20:16.25`	`DeLmAr`	`n\|cotine, well mostly people are all over the place, and each client site is different but most of the time a user is going to be behind nat of some sort.`
`20:17.04`	`DeLmAr`	`n\|cotine, most of the recent model routers seem to support SIP devices quite nicely`
`20:17.11`	`DeLmAr`	`n\|cotine, what issues are you having?`
`20:17.16`	`n\|cotine`	`Not having any yet.`
`20:17.18`	`n\|cotine`	`Still in planning.`
`20:17.49`	`n\|cotine`	`Plan on putting the phones on private IPs - the 3 sites are linked by private lines, so we can route the private IPs internally`
`20:17.52`	`Rhizome`	`Just remember that fax might not work on a sip enabled router`
`20:18.08`	`n\|cotine`	`Just thinking what kind of problems I'd have when adding external phones to the mix`
`20:18.28`	`DeLmAr`	`n\|cotine where is the Asterisk box?`
`20:18.35`	`n\|cotine`	`My first thought was to just set canreinvite to "no" for any external phones, as the asterisk machines will be on both the internal and external networks`
`20:18.45`	`DeLmAr`	`n\|cotine it really does help is the Asterisk box has a public IP`
`20:20.24`	`*** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com)`
`20:20.29`	`Dovid`	`morning`
`20:22.35`	`pigpen`	`Hi all, I need a bit of advice for some dialplan "planning"`
`20:22.47`	`*** join/#asterisk JacksLivr (n=JacksLiv@jules.dougstuff.com)`
`20:23.03`	`Qwell`	`~wikis`
`20:23.06`	`jbot`	`well, wikis is http://www.voip-info.org`
`20:23.06`	`JacksLivr`	`hello all. I just installed *now this morning. I set up the outgoing port to be port 4 (FXO) of my 4 port card. I noticed that when I rang in, it came in on port 4, but when i ring out, the channel status screen showed that it was going out port 3. i moved the phone line over and now it rings out`
`20:23.06`	`pigpen`	`I am working with * 1.4, with RTA on postgres.`
`20:23.08`	`Dovid`	`pigpen: whats up ?`
`20:23.08`	`Dovid`	`lol`
`20:23.08`	`Qwell`	`pigpen: any time`
`20:23.12`	`pigpen`	`hey.`
`20:23.42`	`pigpen`	`Should I keep my main extension handling (like forwarding, dnd, followme, etc...) in the extensions.conf or put it in postgres.`
`20:24.04`	`pigpen`	`the catch is that with RTA with a database backend, you cannot have variables.`
`20:24.19`	`JacksLivr`	`should/can I just close the browser and use the asterisknow install with just the config files. I'm a n00b.`
`20:24.40`	`JacksLivr`	`or is it better to build a fedora box and start from scratch?`
`20:24.58`	`pigpen`	`ie: if I have a dailplan where when calling an extention it looks for 10 differenet things, I would have in the database 10 differetn things per extensions. (sorry for the spelling)`
`20:25.01`	`pigpen`	`typing fast.`
`20:25.23`	`pigpen`	`Where as in extensions.conf, one set covering them all.`
`20:25.59`	`pigpen`	`I hope I made some sense.`
`20:27.29`	`pigpen`	`I am thinking, keep the "mass variable stuff" in the extensions.conf, putting extension specific in the database.`
`20:27.51`	`Defend`	`hey fellas any one happen to know where the new asterisk gui puts log info? i goto log in and it imedately logs me out so i was looking for a log file to see what is going on`
`20:28.46`	`Bobthehunter`	`so wahts up with fromuser....`
`20:28.52`	`Bobthehunter`	`its desteroying call from info`
`20:29.11`	`pigpen`	`Defend, sorry, I've never worked with it.`
`20:29.31`	`Defend`	`alright thanks tho mate :/`
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`20:29.40`	`bkervaski`	`Greets.`
`20:31.12`	`pigpen`	`shit..I think I ran everyone off.`
`20:31.21`	`pigpen`	`:)`
`20:31.37`	`JacksLivr`	`can anyone read this?`
`20:31.44`	`pigpen`	`read waht?`
`20:31.51`	`pigpen`	`s/waht/what`
`20:31.59`	`JacksLivr`	`ok, thanks. i wasnt sure everyone could see me`
`20:32.03`	`bkervaski`	`No.`
`20:32.07`	`pigpen`	`see what?`
`20:32.29`	`JacksLivr`	`i didnt know if i had logged on to freenode... oh`
`20:32.38`	`pigpen`	`yeah..you made it.`
`20:33.06`	`JacksLivr`	`should i punt *now and start over with fedora, etc?`
`20:33.38`	`pigpen`	`start over with Gentoo.`
`20:33.41`	`bkervaski`	`Ubuntu 6.06LTS Server.`
`20:33.59`	`Defend`	`lol!!!`
`20:34.01`	`JacksLivr`	`ive fun fedora and freebsd, not really the others`
`20:34.04`	`pigpen`	`distro war...`
`20:34.07`	`bkervaski`	`lol`
`20:34.08`	`Defend`	`the asterisk gui doesnt work with ie`
`20:34.17`	`JacksLivr`	`im using firefox`
`20:34.18`	`Defend`	`i swaped to ff and it works`
`20:34.46`	`JacksLivr`	`i can do stuff int he gui, its just seems buggy`
`20:35.25`	`Defend`	`i just want it for quick start so i can let people make calls while i read docs`
`20:35.35`	`*** join/#asterisk tom_kelleher (n=Tom@adsl-71-141-224-101.dsl.snfc21.pacbell.net)`
`20:36.30`	`tom_kelleher`	`hello, I am curious if it is possible to use Asterisk to create a "virtual conferance room" on the server`
`20:36.50`	`pigpen`	`tom_kelleher, meetme..yes.`
`20:37.19`	`tom_kelleher`	`pigpen, could you please breifly explain how that works.`
`20:38.33`	`pigpen`	`Well, you call into an exten and punch in a pin if necessary, and you and whomever is in a conf room.`
`20:38.34`	`pigpen`	`http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe`
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`20:38.47`	`pigpen`	`lots of options.`
`20:39.17`	`pigpen`	`One of my customers use it when they have group training.`
`20:39.18`	`tom_kelleher`	`thank you pigpen.`
`20:39.41`	`pigpen`	`one person calls the training facility, transfers the call to the conf room...then the other 60 people joins in.`
`20:40.44`	`*** join/#asterisk digiportbram (n=bram@72-254-136-136.client.stsn.net)`
`20:40.50`	`*** join/#asterisk mikeekim (n=mike@204.13.2.6)`
`20:40.54`	`mikeekim`	`hello ladies`
`20:41.22`	`digiportbram`	`anyone using accountcode on a per user basis in iax.conf?`
`20:41.50`	`DeLmAr`	`digiportbram yup`
`20:41.51`	`digiportbram`	`seems to have a bug that makes it pickup the last accountcode= setting in iax.conf instead of the ones based on user`
`20:42.08`	`DeLmAr`	`digiportbram cant say im having that problem.`
`20:42.20`	`digiportbram`	`delmar: hmm...`
`20:42.28`	`digiportbram`	`delmar: 1.4.0?`
`20:42.36`	`DeLmAr`	`digiportbram no. 1.2.15`
`20:42.39`	`digiportbram`	`hmm`
`20:42.53`	`digiportbram`	`delmar: i really don't want to downgrade`
`20:43.16`	`file`	`did you confirm that the correct user entry is being matched?`
`20:43.49`	`digiportbram`	`well...not sure if it matters, but the are friends not users`
`20:43.50`	`DeLmAr`	`digiportbram, ill be damned if im going to upgrade.. just yet :P. too many other things to do than replace something with something else when the first something aint broke, and the upgrade would mean I have to actually do some work to make sure configs confirm and such like :P`
`20:43.51`	`*** join/#asterisk ocgltd (n=support@CPE004063e0ee74-CM00159a010632.cpe.net.cable.rogers.com)`
`20:43.56`	`DeLmAr`	`rather waste time on something else.`
`20:43.57`	`DeLmAr`	`:)`
`20:44.10`	`digiportbram`	`right`
`20:44.25`	`ocgltd`	`Does anyone have experience with OPAL and Ast 1.40 ?`
`20:44.34`	`DeLmAr`	`might take a look at it in the next few weeks tho.`
`20:44.51`	`file`	`ocgltd: chan_h323 does not have the capability to use OPAL`
`20:44.55`	`*** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net)`
`20:45.02`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`20:45.08`	`DeLmAr`	`file, hey man. long time.`
`20:45.09`	`digiportbram`	`thanks delmar`
`20:45.23`	`digiportbram`	`maybe downgrade`
`20:46.04`	`digiportbram`	`which leads me to my next question then...anyone know is 1.2.x and 1.4.0 have interop issues?`
`20:46.09`	`digiportbram`	`if^`
`20:46.32`	`file`	`digiportbram: if you are using SIP with RFC2833 for DTMF between them you need to turn on an option, but besides that no - shouldn't be any issues`
`20:46.49`	`ocgltd`	`I have integrated Open H323 1.18 into Ast 1.4. Although it runs, I've run into a problem. I'm trying to send a t.38 fax from leg (H323) to another leg (SIP).`
`20:46.54`	`digiportbram`	`thanks file`
`20:47.00`	`DeLmAr`	`digiportbram, I was under the impression that there were a few differences in configs and you would need to make changes to youre 1.2.x configs when going up to 1.4 so it would stand to reason you would need to make changes to go back..`
`20:47.20`	`DeLmAr`	`file, how much is different between them that needs changing ?`
`20:47.20`	`digiportbram`	`i would be doing iax sessions between several * servers`
`20:47.28`	`ocgltd`	`The problem is that the connection drops. Although voice calls pass through fine, t.38 calls fail`
`20:47.38`	`file`	`DeLmAr: depends... but it is all detailed in the UPGRADE filee`
`20:47.51`	`digiportbram`	`thanks again delmar, file`
`20:47.55`	`digiportbram`	`:)`
`20:48.01`	`file`	`ocgltd: T.38 isn't supported in chan_h323, so you would have to probably rewrite how T.38 is done to add support`
`20:48.35`	`Qwell`	`file: he was told that twice earlier`
`20:48.38`	`DeLmAr`	`cheers everyone. back later perhaps.`
`20:48.55`	`*** part/#asterisk tom_kelleher (n=Tom@adsl-71-141-224-101.dsl.snfc21.pacbell.net)`
`20:49.02`	`file`	`Qwell: excellent!`
`20:49.49`	`Qwell`	`skinny_set_lamp(s, sd->instance, SKINNY_LAMP_LAVA);`
`20:50.13`	`Qwell`	`I'm gonna do that when I get the 7970 working`
`20:51.32`	`bkervaski`	`Qwell: You guys use Cisco phones, right? In an Asterisk 1.2.x install, where NAT is required, any suggestions on how to make the Cisco 79x1's NAT friendly with Asterisk? We're all out of things to try.`
`20:51.36`	`bkervaski`	`SIP, not skinny.`
`20:51.52`	`Qwell`	`no`
`20:52.03`	`Qwell`	`SIP sucks`
`20:52.08`	`Qwell`	`Cisco SIP moreso`
`20:52.08`	`bkervaski`	`Ok, that's where we were.`
`20:52.09`	`*** join/#asterisk DrukenLPY (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com)`
`20:52.41`	`bkervaski`	`We can't upgrade to 1.4 yet, too many users to risk it. I guess we're just stuck for now. The 1.2.x branch doesn't recognize the 79x1's.`
`20:53.08`	`*** join/#asterisk uNK_ (i=uNK@modemcable246.26-70-69.static.videotron.ca)`
`20:53.27`	`bkervaski`	`(for chan_skinny)`
`20:53.42`	`Qwell`	`I wouldn't use chan_skinny in 1.2 anyhow`
`20:53.56`	`Qwell`	`not unless Corydon-w releases his backport`
`20:54.01`	`ocgltd`	`File/Qwell: I've gone back to Open 323 site and from what I understood, their H.323 implementation supports T.38. I thought (perhaps mistakenly) that this replaces chan_h323. Am I mistaken here?`
`20:54.47`	`bkervaski`	`Ok. Oh well, when 1.4.1 official hits we'll throw it in the lab. Thanks, Qwell.`
`20:54.47`	`*** join/#asterisk drako (n=ljd@unaffiliated/luisjose)`
`20:54.47`	`bkervaski`	`That weather hit you guys yet? (We're down in Birmingham).`
`20:54.48`	`Qwell`	`it's pretty windy here`
`20:55.07`	`bkw__`	`man this is cool http://www.phonetrace.org`
`20:55.23`	`file`	`ocgltd: OpenH323 is a library that chan_h323 uses for H323 capability, chan_h323 would still need to support T.38 and it does not`
`20:55.35`	`Qwell`	`bkw__: apparently I can find a real sex partner in Huntsville - now!`
`20:55.45`	`bkervaski`	`That's just wrong, lol.`
`20:56.06`	`bkw__`	`haha`
`20:56.20`	`Qwell`	`bkw__: does it...work?`
`20:56.34`	`Qwell`	`and, btw, did you read the little note at the bottom?`
`20:56.41`	`Qwell`	`"we may contact you in the future with offers and promotions"`
`20:56.42`	`pigpen`	`bkw__, playing for the other team?`
`20:56.46`	`nibbler_de`	`how did they get this bedroom pic of my place here`
`20:57.16`	`Qwell`	`rofl`
`20:57.20`	`uNK_`	`there we go`
`20:57.23`	`Qwell`	`oh god`
`20:57.24`	`uNK_`	`was waiting for you to try it`
`20:57.25`	`uNK_`	`hehe`
`20:57.43`	`uNK_`	`my fav web service ;)`
`20:57.43`	`Defend`	`any idea what this mean? Correct auth, but based on stale nonce received from`
`20:57.46`	`bkw__`	`pigpen, gee you figure that one out your own?`
`20:58.00`	`ocgltd`	`file: ok I'm getting it now. Is there another way to achieve this (T.38 passthrough from h323 to sip) in Asterisk?`
`20:58.10`	`ocgltd`	`Or do I have to look at another product?`
`20:58.32`	`pigpen`	`:p`
`20:58.51`	`*** join/#asterisk LeeEMel (n=lee@kereedco.com)`
`20:58.59`	`Corydon76-home`	`Qwell: the backport isn't quite there yet`
`20:59.00`	`file`	`ocgltd: you will have to look elsewhere`
`20:59.10`	`LeeEMel`	`hello all, anybody ever connect an asterisk box to a shoretel system using sip?`
`20:59.26`	`Qwell`	`Corydon76-home: see my patches from last night :)`
`20:59.54`	`Corydon76-home`	`Mmm?`
`20:59.59`	`Qwell`	`devicestate ;)`
`21:00.05`	`Qwell`	`(and a couple bug fixes)`
`21:00.25`	`LeeEMel`	`i semm to beable to register just fine. I can call out just fine but an incoming call gives me handle_request_invite: Failed to authenticate user`
`21:00.45`	`bkervaski`	`Anyone run SIP/RTP through an IPSEC vpn? If so, how does it perform? Is the extra overhead noticeable?`
`21:00.52`	`nibbler_de`	`does anbody of you use the siemens gigaset 450IP?`
`21:01.39`	`pigpen`	`bkervaski, yes...works better in my case, as my isp has a higher QOS for vpn traffic over sip.`
`21:01.57`	`ocgltd`	`file: I think that I read that future versions of asterisk will use OPAL, and that chan_h323 dev is stopped. Anyone up on this?`
`21:02.06`	`bkervaski`	`pigpen: Thanks!`
`21:02.30`	`Qwell`	`ocgltd: there's been talk of compiling against opal`
`21:02.30`	`file`	`ocgltd: Paul Cadach heads up chan_h323 stuff these days, and he made a note of that... but it's up to him`
`21:03.29`	`*** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)`
`21:03.37`	`*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)`
`21:04.21`	`*** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)`
`21:06.44`	`*** join/#asterisk ruied (n=ruied@bl7-218-228.dsl.telepac.pt)`
`21:11.40`	`*** join/#asterisk InHisName (n=Administ@c-68-38-105-1.hsd1.pa.comcast.net)`
`21:12.52`	`InHisName`	`anyone working with testfeature command ?`
`21:12.53`	`pigpen`	`Am I correct that all dialplan statements stored in databases (mine is in postgresql), commas "," are not allowed, use " \| " instead?`
`21:13.34`	`*** join/#asterisk dahunter3 (n=dahunter@pool-71-177-150-211.lsanca.fios.verizon.net)`
`21:13.57`	`InHisName`	`pigpen could be as many store databases as csv files.`
`21:14.16`	`*** part/#asterisk dahunter3 (n=dahunter@pool-71-177-150-211.lsanca.fios.verizon.net)`
`21:14.26`	`Bobthehunter`	`supertec.crap..`
`21:14.31`	`Bobthehunter`	`hickaing domains again`
`21:14.36`	`Bobthehunter`	`highjacking i mean`
`21:14.58`	`pigpen`	`I read a note that in GoTo and Dial, use \| instead of , ....one way to find out.`
`21:15.10`	`pigpen`	`hehe.`
`21:15.23`	`*** join/#asterisk yassine (n=yassine@dsl.voicint.com)`
`21:16.28`	`Defraz`	`would playtones have any issues with using g729?`
`21:16.30`	`Defraz`	`I wouldn't think so.`
`21:16.43`	`Defraz`	`I just can't get the busy tone to generate.`
`21:18.57`	`nibbler_de`	`Defraz: no possibility to signal busy? would be much cleaner if you ask me.`
`21:19.50`	`Defraz`	`I do after I answer I playtone busy then I use the busy() function`
`21:20.19`	`nibbler_de`	`why the busy tone at all?`
`21:20.52`	`Defraz`	`cuz figured it would help`
`21:21.23`	`*** join/#asterisk darken_darken (n=marco@239.143.76.83.cust.bluewin.ch)`
`21:21.30`	`Defraz`	`haha`
`21:21.34`	`Defraz`	`and it said it wouldn't hurt`
`21:21.42`	`Defraz`	`on the Wiki on what I have found.`
`21:22.32`	`nibbler_de`	`help (with) what?`
`21:23.02`	`Defraz`	`playing the tone`
`21:23.22`	`*** join/#asterisk daveheun (i=davidheu@196.211.34.2)`
`21:23.31`	`daveheun`	`yes\yes`
`21:23.32`	`Defraz`	`said some providers don't play a tone when they receive the busy() notification`
`21:23.36`	`Defraz`	`so it can't hurt`
`21:23.40`	`daveheun`	`hi there people`
`21:23.43`	`InHisName`	`Defraz = answers, plays tone, then sets busy(), is that right ?`
`21:24.02`	`nibbler_de`	`why should they play a tone? they pass the information "busy" to the other party, done/done`
`21:24.02`	`Defraz`	`yes`
`21:24.32`	`Defraz`	`Well, nibbler_de I agree with you but, just was trying what the wikii said or other forums.`
`21:24.47`	`Defraz`	`It is always my last resort to ask in here. Been looking all over.`
`21:24.47`	`nibbler_de`	`the equipment of the calling/called party is responsible to signal "busy"`
`21:25.03`	`InHisName`	`Defraz to what purpose are these 3 steps do for you ?`
`21:25.15`	`Defend`	`i am liken 1.4 so far`
`21:25.36`	`daveheun`	`anyone have some time to help me with a problen an a asterisk system with b410p isdn module`
`21:25.38`	`Defraz`	`If the SIP device is busy with a call.`
`21:25.43`	`Defraz`	`then it will ring busy.`
`21:26.01`	`Defraz`	`Unless I direct it to something else. like voicemail or something.`
`21:26.09`	`InHisName`	`Hmmm, I might need that too.`
`21:26.48`	`InHisName`	`Right now I am trying to get testfeature to work first then transfers to work.`
`21:27.17`	`daveheun`	`anyone have a sample config file for extensions.conf to answer a call from misdn.conf?`
`21:28.07`	`nibbler_de`	`exten => 3000,1,Answer`
`21:28.47`	`daveheun`	`i have exten => _X.,1,Playback(vm-goodbye) to test but dont get error: cant match extention`
`21:28.53`	`*** join/#asterisk labadaba (i=labadaba@83.243.88.163)`
`21:28.57`	`labadaba`	`#Funny #Funny #Funny #Funny #Funny #Funny #Funny #Funny #Funny`
`21:29.02`	`*** part/#asterisk labadaba (i=labadaba@83.243.88.163)`
`21:29.23`	`daveheun`	`i have exten => _X.,1,Playback(vm-goodbye) to test but get error: cant match extention`
`21:30.20`	`daveheun`	`l`
`21:30.22`	`*** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C)`
`21:30.33`	`daveheun`	`anyone active?`
`21:30.33`	`DocHolliday`	`anyone done anything interesting with the 7941?`
`21:30.46`	`nibbler_de`	`yeah.`
`21:31.08`	`nibbler_de`	`i made some...`
`21:31.47`	`nibbler_de`	`phonecall!`
`21:31.47`	`daveheun`	`rules`
`21:31.48`	`daveheun`	`!rules`
`21:33.21`	`daveheun`	`can anyone help me please`
`21:34.17`	`*** part/#asterisk daveheun (i=davidheu@196.211.34.2)`
`21:34.29`	`*** join/#asterisk daveheun (i=davidheu@196.211.34.2)`
`21:34.44`	`sweeper`	`daveheun: make sure your contexts are correct`
`21:35.38`	`daveheun`	`sweeper: i am sure it is rigth`
`21:36.05`	`sweeper`	`daveheun: pastebin logs plz`
`21:38.59`	`daveheun`	`misdn.conf [incoming] ports=1,2 msns=* context=misdn-in ***** extensions.conf [misdn-in] exten => _X.,1,Playback(vm-goodbye}`
`21:39.25`	`daveheun`	`can,t paste logs i am at home now`
`21:39.42`	`daveheun`	`have this problem at home`
`21:39.49`	`daveheun`	`sorry work`
`21:42.03`	`*** join/#asterisk ghenry (n=ghenry@87.112.16.56.plusnet.ptn-ag1.dyn.plus.net)`
`21:45.07`	`InHisName`	`duplicate: exten => _X.,1,Playback(vm-goodbye) and change the _X to 's'. Change vm-goodbye to tt-monkeysintro, then try test again.`
`21:47.52`	`InHisName`	`daveheun, may want to setup extensions i, t also just to find which extension it is trying.`
`21:47.52`	`daveheun`	`explain please`
`21:47.52`	`*** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep)`
`21:47.52`	`daveheun`	`i did thy the s,1,Answer() and then exten => _X.,1,Playback(vm-goodbye}`
`21:48.01`	`daveheun`	`i did try the s,1,Answer() and then exten => _X.,1,Playback(vm-goodbye}`
`21:48.11`	`InHisName`	`copy the test line 3 more times and change the "_X" to: 1. s, 2. i, 3. t then change the mesage to 3 others then test the call.`
`21:48.55`	`InHisName`	`daveheun: still get the error: get error: cant match extention ?`
`21:49.17`	`daveheun`	`yip`
`21:49.54`	`InHisName`	`daveheun are you watching the CLI> while testing ? What is verbose level ?`
`21:50.05`	`Bobthehunter`	`oh my god.. rehan and is trolls are just..`
`21:50.15`	`daveheun`	`what does the 1.s and the rest do?`
`21:50.21`	`Bobthehunter`	`he puts clients lists as they where clients when its all is troll sub sites...`
`21:50.25`	`daveheun`	`verbose 10`
`21:50.25`	`Bobthehunter`	`of is`
`21:51.08`	`InHisName`	`just s i t s is default, i is for error conditions and t for timeout conditions.`
`21:52.44`	`InHisName`	`if you have dialed a number then _X. is best match. s is most useful for incoming calls that have not dialed any extensions yet.`
`21:53.12`	`InHisName`	`i is when they dialed and error encountered and EXTEN is set to i`
`21:53.20`	`daveheun`	`ohhh i see you meed s,1,Playback(vm-hello) i,1,Playback(vm-test) t,1,Playback(vm-goodbye)`
`21:53.56`	`InHisName`	`yes, then check for error. OR you may be in different context that what you thought.`
`21:54.58`	`InHisName`	`Seeing the CLI output with high verbose settings can clue you in which context you are using when a number is dialed by user.`
`21:55.20`	`daveheun`	`yip if i set the misdn set debug to 5 i cann see a lot of info running down the screen what does oad: 8007 mean`
`21:56.06`	`daveheun`	`8007 is the extention i am dialing from from is 8007 so is that the did number`
`21:57.04`	`*** join/#asterisk vgster (n=vgster@81.96.139.59)`
`21:57.16`	`daveheun`	`i then tried doing something like this exten => 8007,1,Dial(SIP/100)`
`21:58.01`	`daveheun`	`to test but still error: cantmatchext`
`21:58.11`	`DocHolliday`	`anyone here have a cisco 7941/61?`
`21:58.32`	`InHisName`	`exten => s,n,NoOp(user=${EXTEN}) will leave marks in CLI so you can see where things are heading and values of var are like.`
`21:58.33`	`jql`	`yeah, I have a 7961`
`21:58.36`	`*** join/#asterisk vgster (n=vgster@81.96.139.59)`
`21:59.12`	`*** join/#asterisk Piano_ (n=Piano@c-67-175-92-171.hsd1.il.comcast.net)`
`21:59.52`	`InHisName`	`Change the _X,1 to _X,2 and do the NoOp as _X.,1 or try earlier where you know it runs.`
`22:00.22`	`*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)`
`22:00.33`	`daveheun`	`tried NoOp does not write screen`
`22:00.54`	`jql`	`probably because of the context. hmm`
`22:01.03`	`daveheun`	`got noop to work on fxo ports not the b410p diguim module`
`22:02.01`	`*** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il)`
`22:03.53`	`daveheun`	`is this an australian channel?`
`22:03.58`	`pigpen`	`Didn't I see a patch that would allow the asterisk dialplan to access items in a postgres database?`
`22:04.05`	`*** join/#asterisk ruied (n=ruied@bl7-218-228.dsl.telepac.pt)`
`22:04.09`	`pigpen`	`ie: instead of the built in db?`
`22:04.13`	`DocHolliday`	`jql, running SIP firmware?`
`22:04.22`	`*** join/#asterisk th3073ch (n=stephen@71.14.141.169)`
`22:04.22`	`jql`	`DocHolliday: yep`
`22:04.32`	`DocHolliday`	`jql, what version of SIP are you running?`
`22:04.43`	`daveheun`	`anyone check the rugby today?`
`22:04.51`	`*** join/#asterisk ghenry (n=ghenry@87.112.16.56.plusnet.ptn-ag1.dyn.plus.net)`
`22:04.57`	`jql`	`good question. let me see`
`22:05.06`	`DocHolliday`	`thanks :)`
`22:05.25`	`Dovid`	`nope`
`22:05.31`	`Dovid`	`its a worldwide one ;)`
`22:05.42`	`daveheun`	`ok`
`22:05.45`	`DocHolliday`	`daveheun, i should.. but im busy :(`
`22:05.46`	`daveheun`	`dovid`
`22:05.58`	`Dovid`	`dont know much - but can try to hel`
`22:05.59`	`Dovid`	`help*`
`22:06.01`	`Dovid`	`whats up?`
`22:06.24`	`daveheun`	`scrathing my head with asterisk`
`22:06.35`	`daveheun`	`you anygood on asterisk`
`22:06.36`	`jql`	`8-0-2SR1-0-1 is what the loads file says`
`22:06.37`	`daveheun`	`?`
`22:07.01`	`DocHolliday`	`jql, is the phone stable?`
`22:07.19`	`jql`	`been running that revision for over 9 months without issue`
`22:07.57`	`DocHolliday`	`great, i have noticed that with the 79x1 SIP the call forward button is missing, have you found a work around?`
`22:08.28`	`jql`	`Hmm. The call forward button is a soft-key`
`22:08.36`	`jql`	`and it works fine with asterisk`
`22:08.38`	`DocHolliday`	`right :)`
`22:08.46`	`DocHolliday`	`oh its actually there on your phone!??!`
`22:08.49`	`jql`	`yes`
`22:09.03`	`jql`	`used it for testing`
`22:09.40`	`DocHolliday`	`i am runing 8-2-0-55 and that button is missing`
`22:09.52`	`jql`	`well... remind me not to use that revision`
`22:10.24`	`DocHolliday`	`hah, on your version does call transfer work?`
`22:10.37`	`jql`	`Yes`
`22:10.45`	`LeeEMel`	`any shoretel users out there`
`22:10.48`	`jql`	`strangely enough, it's the Polycom that I have trouble transferring with`
`22:10.49`	`daveheun`	`does anyone know ehere i can get some docs on linking misd.conf to extensions.conf???????????`
`22:10.56`	`jql`	`the call transfers, but the Polycom doesn't hang up`
`22:11.02`	`jql`	`still working on that...`
`22:11.26`	`DocHolliday`	`i have the same problem but with this cisco phone :P`
`22:12.23`	`DocHolliday`	`the call transfers but my phone holds on to the call in a 'hold' position`
`22:12.23`	`jql`	`yeah, same for me with the Polycom. Very weird`
`22:12.25`	`KnowWhat`	`i am getting this problem`
`22:12.25`	`KnowWhat`	`error in make /usr/include/linux/ixjuser.h:351: error: syntax error before '*' token`
`22:12.25`	`DocHolliday`	`jql, is there a chance you could give a copy of the firmware you are using? please`
`22:12.33`	`KnowWhat`	`what could be the problem`
`22:14.45`	`*** join/#asterisk hads (n=hads@reef80.anchor.net.au)`
`22:15.06`	`Defend`	`any one here fimilar with jitter control on 1.4?`
`22:20.22`	`Hmmhesays`	`stone cold crazy is such a bitch to play on guitar`
`22:20.48`	`*** join/#asterisk olsen (n=diego@200.61.236.33)`
`22:21.30`	`kanaeda`	`lols`
`22:21.35`	`kanaeda`	`thats old school`
`22:21.59`	`Hmmhesays`	`I'm learning the metallica remake version`
`22:22.08`	`kanaeda`	`remake?`
`22:22.36`	`Hmmhesays`	`metallica covered the song`
`22:23.02`	`KnowWhat`	`error in make /usr/include/linux/ixjuser.h:351: error: syntax error before '*' token`
`22:25.11`	`*** join/#asterisk daveheun (i=davidheu@196.211.34.2)`
`22:26.57`	`*** join/#asterisk dseeb_ (n=dcb@CPE-124-177-40-105.vic.bigpond.net.au)`
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`22:37.13`	`InHisName`	`anyone working with testfeature command ?`
`22:44.38`	`DocHolliday`	`anyone been able to get a do not disturb softkey on their Cisco 7941/61?`
`22:47.52`	`bkervaski`	`DocHolliday: Not me, but would like one :)`
`22:48.15`	`DocHolliday`	`hey bkervaski, thanks for helping me before :)`
`22:48.37`	`DocHolliday`	`LostFrog, Tor = Latency ++`
`22:48.38`	`bkervaski`	`np`
`22:49.06`	`DocHolliday`	`bkervaski, the firmware works its just that for example when i transfer a call.. it transfers but holds on to it :P`
`22:49.24`	`bkervaski`	`transfers but holds on to it? What do you mean?`
`22:49.52`	`DocHolliday`	`the cisco phone initiating the transfer will transfer the call put it will stay on the phone in a 'hold' status`
`22:50.15`	`bkervaski`	`That's odd. Mine doesn't. hrm...`
`22:50.16`	`DocHolliday`	`*but`
`22:50.34`	`*** part/#asterisk th3073ch (n=stephen@71.14.141.169)`
`22:51.01`	`DocHolliday`	`yeah`
`22:51.29`	`DocHolliday`	`it doesnt even transfer actually.. it just places the transferred phone in a hold status, and the call transfer initiator in a hold status :P`
`22:51.40`	`bkervaski`	`So the Cisco phone is basically bridging the call?`
`22:51.42`	`*** join/#asterisk bmd (n=bmd@72.54.252.34)`
`22:51.57`	`DocHolliday`	`bkervaski, as per my analysis right now it just puts both phones on hold`
`22:52.34`	`bkervaski`	`Weird. Our transfers work fine... Dunno on that one.. But definitely let me know if you fix it and what you did..`
`22:53.18`	`DocHolliday`	`will do :)`
`22:53.38`	`DocHolliday`	`bkervaski, but you have been unsuccessful in getting a do not disturb button?`
`22:54.35`	`bkervaski`	`I didn't spend a lot of time on it, but our polycoms have it and I like the feature.`
`22:56.13`	`DocHolliday`	`gotcha :)`
`22:57.09`	`*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)`
`22:57.47`	`DocHolliday`	`bkervaski, this what i just determined.. the 'transfer' function doesnt work, but if i dial my Cisco 7940 from my 7941 and do a 'blind transfer' it works`
`22:58.27`	`*** join/#asterisk logicwrath (n=some@c-68-60-121-112.hsd1.mi.comcast.net)`
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`22:58.30`	`*** join/#asterisk weazahl (n=weazahl@adsl-66-136-148-13.dsl.kscymo.swbell.net)`
`22:58.40`	`bkervaski`	`Ahh... So you're calling the transferee, then hitting transfer... That's just a conference call and your phone is bridging...`
`22:59.28`	`weazahl`	`im having problems calling from one GS386 port to another... is this the problem? 6002/6002 192.168.1.196 D N 5062 OK (6 ms)`
`22:59.36`	`DocHolliday`	`bkervaski, Cisco 7941 -- > Cisco 7940 -- > [7940 transfers to voicemail] .. puts both phones on hold`
`22:59.38`	`weazahl`	`port 5062??????`
`22:59.46`	`bkervaski`	`Hrm...`
`23:00.07`	`bkervaski`	`(Sorry, I'm half paying attention to this, trying to get a damn IPSEC tunnel to work on an Ubuntu box)`
`23:00.21`	`DocHolliday`	`bkervaski, Cisco 7941-->Cisco 7940 --> [BlindXFR to voicemail] = WORkS`
`23:00.37`	`DocHolliday`	`unfortunately my cisco 7941 does NOT have the BlindXFR softbutton`
`23:00.55`	`*** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk)`
`23:01.40`	`weazahl`	`i never saw anything try to register to port 5062 what is up with that?`
`23:01.53`	`nibbler_de`	`DocHolliday: which software do you run on the 7941?`
`23:02.08`	`bkervaski`	`8.2.1 for the 79x1's`
`23:02.29`	`DocHolliday`	`8-2-0-55`
`23:02.35`	`*** join/#asterisk bmg505 (n=leon@c1-134-1.rndf.isadsl.co.za)`
`23:02.38`	`nibbler_de`	`hmm, ok. reasonably recent.`
`23:02.45`	`nibbler_de`	`strange that there's no blindxfer button`
`23:02.46`	`DocHolliday`	`bkervaski, nope 8-2-0 not 8-2-1`
`23:02.59`	`*** join/#asterisk AJaymn (i=AJaymn@24-159-236-181.dhcp.mdsn.wi.charter.com)`
`23:03.00`	`DocHolliday`	`nibbler_de, maybe im missing a line in my SEP file?`
`23:03.05`	`bkervaski`	`nibbler_de: Do you have one on yours?`
`23:04.03`	`DocHolliday`	`bkervaski, are you running 8-2-1 or 8-2-0 ? :P`
`23:04.09`	`bkervaski`	`THe one I sent you...`
`23:04.13`	`bkervaski`	`It's the most recent...`
`23:04.18`	`DocHolliday`	`which is 8-2-0 :)`
`23:04.46`	`bkervaski`	`I stand corrected, sir.`
`23:05.12`	`DocHolliday`	`no problem :)`
`23:05.12`	`daveheun`	`In his name i think i have it check msns`
`23:05.13`	`daveheun`	`Calls whose destination is equal to this will be ignored. This doesn't make sense to me and I suspect this is a relict from an earlier version..`
`23:05.59`	`DocHolliday`	`bkervaski, hah i just solved the DND problem`
`23:06.05`	`daveheun`	`one more question pls`
`23:06.17`	`DocHolliday`	`you have to enable it through the menus`
`23:07.49`	`bkervaski`	`Cool! What option? If it's a menu option then its a config xml option as well...`
`23:08.33`	`DocHolliday`	`Settings --> Device Config --> Call Preferences`
`23:08.41`	`DocHolliday`	`issue is when i enable it and then click the soft key it disappears :)`
`23:09.28`	`daveheun`	`in your misdn.conf file you have a setting where you can select msns=??? i have this set to msns=* on this site on misdn it says "calls whose destination is equal to this will be ignored" saw this on http://www.voip-info.org/wiki/view/Asterisk+config+misdn.conf`
`23:09.47`	`DocHolliday`	`bkervaski, it auto enables.. when you press it in (it disables), but then its gone :P`
`23:09.57`	`daveheun`	`might this be the reason my calls are ignored???????`
`23:10.46`	`*** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it)`
`23:11.12`	`daveheun`	`InHisName are active?`
`23:11.17`	`*** join/#asterisk ghenry (n=ghenry@87.112.16.56.plusnet.ptn-ag1.dyn.plus.net)`
`23:11.23`	`daveheun`	`InHisName are u active?`
`23:11.34`	`bkervaski`	`DocHolliday: How did you change the setting? Mine says it's locked?`
`23:11.48`	`InHisName`	`daveheun i am here. Cant get win popup to work right.`
`23:12.04`	`DocHolliday`	`bkervaski, you have to unlock your phone first.`
`23:12.10`	`DocHolliday`	`settings --> **#`
`23:12.45`	`daveheun`	`can i ask you one last qeustion please`
`23:12.55`	`daveheun`	`?`
`23:12.58`	`bkervaski`	`Yea, did that. But it won't let me change it.. hrm..`
`23:13.00`	`InHisName`	`dhoot`
`23:13.02`	`InHisName`	`shoot`
`23:13.10`	`daveheun`	`<PROTECTED>`
`23:13.41`	`InHisName`	`I am at the site now, reading up on it.`
`23:13.57`	`daveheun`	`thx for your help`
`23:15.34`	`DocHolliday`	`bkervaski, oh`
`23:15.46`	`DocHolliday`	`well thats because you have to set the option to '2' in the configuration`
`23:16.04`	`bkervaski`	`AHh....`
`23:16.17`	`daveheun`	`didnt catch that?`
`23:16.36`	`daveheun`	`what option is that?`
`23:17.20`	`DocHolliday`	`daveheun, me?`
`23:17.43`	`daveheun`	`yes Doc?`
`23:17.45`	`CrazyTux`	`Anyone here have experience with asterisk and AGI type stuff? I.e. say I wanted to make a verify call type setup, where a customer orders something, my system generates a random number, gives them a call, and says the number over the phone, and they have to enter that number into the order system?`
`23:17.59`	`InHisName`	`daveheun msns`
`23:17.59`	`InHisName`	`<PROTECTED>`
`23:18.06`	`DocHolliday`	`<dndControl>2</dndControl>`
`23:18.13`	`DocHolliday`	`but it doesnt really work..`
`23:18.18`	`sweeper`	`CrazyTux: for that, you'd just use call files`
`23:18.21`	`sweeper`	`no need for agi`
`23:18.37`	`CrazyTux`	`sweeper, could you point me to some doc?`
`23:18.52`	`sweeper`	`CrazyTux: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out`
`23:18.58`	`DocHolliday`	`bkervaski, i.e. you can enable it everytime going through the menus.. but not directly`
`23:19.03`	`CrazyTux`	`sweeper, thanks`
`23:19.17`	`sweeper`	`n/o`
`23:19.26`	`*** part/#asterisk daveheun (i=davidheu@196.211.34.2)`
`23:19.44`	`weazahl`	`arrrgh, why cant i make both the ports on my GT 386 ring at the sam time. i can either have one working or the other, not both`
`23:19.46`	`*** join/#asterisk daveheun (i=davidheu@196.211.34.2)`
`23:19.49`	`InHisName`	`daveheun I am really out of my territory ... Reading manual and giving my interpertation is all I can do.`
`23:20.53`	`daveheun`	`thanx for the pointers i will try it tommorrow, thx thx thx may H be with u`
`23:21.22`	`DocHolliday`	`bkervaski, by setting it to '3' it is on automatically but you cant turn it off :P`
`23:21.56`	`*** join/#asterisk voipanywhere (n=pirch@a81-84-60-131.cpe.netcabo.pt)`
`23:23.12`	`InHisName`	`How do I activate features.conf ?`
`23:23.40`	`daveheun`	`just vi or nano it`
`23:23.45`	`jql`	`res_features loaded?`
`23:24.06`	`weazahl`	`i can dial out from my GT386 port 1, but cannot dial into it`
`23:24.11`	`*** join/#asterisk Cresl1n (i=matt@nat/digium/x-4c25e59acfaa4c96)`
`23:24.11`	`*** mode/#asterisk [+o Cresl1n] by ChanServ`
`23:24.11`	`InHisName`	`like this ? reload res_features.so ?`
`23:24.33`	`jql`	`module load res_features.so in certain versions`
`23:24.48`	`jql`	`probably the fantasy version that doesn't exist`
`23:24.53`	`InHisName`	`wait I'll try that....`
`23:25.15`	`DocHolliday`	`jql, have you managed to get DND as a softkey that works on/off?`
`23:25.24`	`jql`	`DocHolliday: yes`
`23:25.42`	`DocHolliday`	`and you are running the firmware version you sent to me?`
`23:25.48`	`jql`	`yes`
`23:25.57`	`weazahl`	`i can dial from port 1 to port 2, can dial out on port 1, but cannot ring port 1`
`23:25.58`	`DocHolliday`	`would you mind telling me what you set? <dndControl> to?`
`23:26.06`	`jql`	`gimmie a minute`
`23:26.12`	`DocHolliday`	`thanks again! :)`
`23:26.34`	`InHisName`	`jql No such command 'module' (type 'help' for help)`
`23:26.52`	`jql`	`InHisName: it might be just 'load' and 'unload`
`23:26.52`	`InHisName`	`jql tried 'load' nope but reload worked at something.`
`23:27.06`	`jql`	`bleh. I'm addicted to asterisk 1.4`
`23:27.16`	`DocHolliday`	`jql, is it worth upgrading?`
`23:27.21`	`jql`	`<PROTECTED>`
`23:27.32`	`jql`	`it had 2 features I required`
`23:27.35`	`jql`	`so I had no choice`
`23:27.44`	`InHisName`	`UNload did something, I guess it is off now ?`
`23:27.54`	`DocHolliday`	`ah, with dndcontrol set to 0 does it appear as a softkey on startup with the ability to turn the feature on/off?`
`23:28.06`	`DocHolliday`	`i.e. once you've pressed it once does it disappear`
`23:28.36`	`jql`	`yes, the softkeys (if I recall -- the phone's on my desk) include Forward and DND in idle mode`
`23:28.47`	`DocHolliday`	`thanks!`
`23:29.02`	`DocHolliday`	`bbl need to reflash this phone :)`
`23:29.14`	`Grnd-Wire`	`jql: What phone are you referring to?`
`23:29.41`	`jql`	`cisco 7961`
`23:29.59`	`bkervaski`	`jql / DocHolliday: What XML config options enable those softkeys?`
`23:30.04`	`bkervaski`	`(I wasn't paying attention)`
`23:30.07`	`DocHolliday`	`oh`
`23:30.14`	`jql`	`Forward is <localCfwdEnable>true</localCfwdEnable>`
`23:30.21`	`jql`	`and dnd was the one above, I believe`
`23:30.44`	`DocHolliday`	`correct`
`23:32.12`	`bkervaski`	`(tnx)`
`23:32.54`	`DocHolliday`	`jql, do you know if there is a way to make the phone accept new firmware without doing a factory reset?`
`23:33.01`	`*** join/#asterisk hematitec (n=cratz@adsl-71-159-206-4.dsl.pltn13.sbcglobal.net)`
`23:33.28`	`jql`	`the .loads file is what determines the firmware revision it loads`
`23:33.35`	`jql`	`to downgrade requires a reset, I believe`
`23:33.38`	`jql`	`upgrades no`
`23:33.45`	`DocHolliday`	`thanks again, heh`
`23:34.07`	`bkervaski`	`DocHolliday: Did you find newer firmware?`
`23:34.25`	`DocHolliday`	`jql, provided me with a copy, i haven't had a chance to test it though`
`23:34.29`	`DocHolliday`	`bbs (dinner)`
`23:34.33`	`bkervaski`	`Which version?`
`23:34.45`	`bkervaski`	`I thought 8.2.0 was the latest...`
`23:34.56`	`jql`	`it's older, not newer`
`23:35.08`	`bkervaski`	`Ahh. Does it work with NAT properly?`
`23:35.21`	`jql`	`no`
`23:35.28`	`bkervaski`	`Oh well :)`
`23:35.41`	`jql`	`Indeed. It works by virtue of the PIX at work`
`23:35.56`	`bkervaski`	`We just did ipsec tunnels with little or no encryption.`
`23:36.06`	`bkervaski`	`Seems to work pretty good actually, surprised.`
`23:36.26`	`bkervaski`	`Fortigates..`
`23:36.27`	`jql`	`the sip-aware PIX firewall opens up the asymmetric SIP ports as needed`
`23:36.46`	`jql`	`which works for anyone beind a PIX. So, not much use to customers`
`23:37.01`	`bkervaski`	`You guys do hosted solutions?`
`23:37.05`	`jql`	`yeah`
`23:37.11`	`bkervaski`	`Which E911 provider?`
`23:37.14`	`jql`	`I do, thay is`
`23:37.18`	`jql`	`intrado`
`23:37.27`	`bkervaski`	`Cool. Same here. What's the name of your company?`
`23:37.35`	`jql`	`I work at FreedomVoice`
`23:38.03`	`jql`	`still under development, so nothing exciting yet`
`23:38.06`	`bkervaski`	`Good deal. HeavyLogic, myself.`
`23:39.23`	`jql`	`I love my Cisco... much more stylish than the Polycom, IMO`
`23:39.45`	`bkervaski`	`Same here. Wish it didn't have so many issues.`
`23:39.47`	`jql`	`And, the linksys/sipura knockoffs don't impress me. I bought one, and it feels cheap`
`23:40.10`	`bkervaski`	`Polycom's are by far the best bang for the buck. If Cisco would get there act together, they would dominate.`
`23:40.29`	`bkervaski`	`Did you guys provision an Introdo solution or are you outsourcing it?`
`23:40.31`	`jql`	`Well, I still have fun with the Polycom nat traversal... they ever fix that?`
`23:40.43`	`jql`	`I handled the Intrado integration personally.`
`23:40.48`	`weazahl`	`on my HT-386 i can dial from port 1 to port 2, can dial out on port 1, but cannot ring port 1. any ideas?`
`23:40.49`	`jql`	`A fun 4 months that was...`
`23:40.54`	`bkervaski`	`Polycom's handle nat no problem.`
`23:41.04`	`jql`	`how do I turn on pinging? heh`
`23:41.50`	`bkervaski`	`Gotta run, ttyl8r`
`23:42.58`	`*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)`
`23:45.08`	`*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)`
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`23:52.56`	`DocHolliday`	`jql, still around?`
`23:53.04`	`jql`	`somewhat`
`23:53.26`	`DocHolliday`	`if i use the version you sent me and its an internal NAT network (asterisk and the phone on the same subnet) will i have a problem?`
`23:54.27`	`jql`	`no`
`23:54.54`	`jql`	`Ciscos use asymmetric SIP, which means sip-ignorane firewalls won't open the incoming port`
`23:54.57`	`jql`	`no firewall, no problem`
`23:55.00`	`DocHolliday`	`OMG i got call transferring work runs around`
`23:55.29`	`jql`	`excellent`
`23:55.31`	`*** join/#asterisk CrashHD (n=crashhd@c-76-20-22-240.hsd1.ca.comcast.net)`
`23:55.42`	`DocHolliday`	`jql, i cant tell whether i got it working in one case or all around`
`23:56.09`	`DocHolliday`	`when i dial from the console i was able to transfer the call from one of the phones to the other phone (so its questionable)`
`23:56.44`	`jql`	`ahh, console. I never really tried that`
`23:58.00`	`DocHolliday`	`jql, yup it works sweetness`
`23:59.59`	`*** join/#asterisk \|dennis\| (n=dennis@200.32.233.82)`

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